Re: [asterisk-users] Inexpensive device for bandwidth management

[hh174]

Mike,

This firmaware works on Buffalo, linksys and some asus routers.

Linksys did release the wrt54gL because of the demand to have a router
with Linux.
In fact, the L means Linux and this router is still in production, easy
to find (in Europe anyway) and very very cheap.

DD-Wrt also runs on it and some other Linux firmware with QOS.

The reason I recommand Tomato is that Tomato is very easy to configure
for end users.
DD-wrt is more complete but also more complex.



Mike a crit:

  
  
  

  
  I
just reread my question and realized I might not have been
clear enough. What I meant is that it only seems to works on older
Linksys
hardware revisions. How do I make sure I can get those?
  
  Mike
  
  
  
  
  From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
  Sent: Sunday, April 05, 2009 15:30
  To: oliv...@hh174.be; 'Asterisk Users Mailing List -
Non-Commercial
Discussion'
  Subject: Re: [asterisk-users] Inexpensive device for bandwidth
management
  
  
  
  Actually
that was my original thought. BUTaccording to
what I read on their FAQ, the hardware that can be used is rather
limited. How do I secure a reliable supply of those?
  
  Mike
  
  
  
  
  
  
  From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hh174
  Sent: Sunday, April 05, 2009 14:49
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Inexpensive device for bandwidth
management
  
  
  
  Linksys (cisco)WRT54GL and the tomato firmware.
  
5 minutes setup
  
Olivier
  
Mike a crit: 
  Thanksthe thing is I need many device (one for each of my hosted
  customers) and I'd like this process to be as easy for non-techies as
  possible, because some of those are technologically-challenged, and need to
  install the box by themselves or with the help of an IT person that only
  knows how to install a run of the mill router.
  
  So an out-of-the-box thing would be better, but I was recommende the pfsense
  before and will take a look at it.
  
  Mike
  
  
   
  
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of drew einhorn
Sent: Sunday, April 05, 2009 13:26
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inexpensive device for bandwidth management

The following two links deal with the same familly of boxes.

Generally it's $20 for a case,
$20 for a powersupply, but you've probably got an old one that will work.
and almost all of their boards are under $200, except for the ones with
lots of gigabit interfaces. Many are under $100.

http://www.mikrotik.com/
http://routerboard.com/

On Sun, Apr 5, 2009 at 11:07 AM, Mike l...@virtutel.ca wrote:
 

  Hi,
  
  
  
  I'm looking for a good network device that does bandwidth management.
   

  
  It
   
  

  can be integrated in a router or stand-alone, but must be SIP-friendly.
  
  
  
  I`ve tried the DIR-655 (latest firmware is SIP-hostile, and the latest
  hardware revisions can't downgrade to the version that worked well) and
   

the
 

  DI-724GU (SIP-friendly, but bandwidth management is automated and not
  configurable enough for my taste), both from D-link.
  
  
  
  What else is out there and allows me to do upstream QoS on cable/DSL
   

links?
 

  Both D-Link routers were under 200$ (99$ and 159$ respectively) and were
  perfect price-wise for my target customers.
  
  
  
  Mike
  
  
  
  
  
  
  
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Re: [asterisk-users] Global h exten

[Dovid Bender]

I had a patch created for 1.4.X for this.
http://bugs.digium.com/bug_view_page.php?bug_id=14159
  - Original Message - 
  From: Gabriel Ortiz Lour 
  To: asterisk-users@lists.digium.com 
  Sent: Wednesday, March 18, 2009 8:23 PM
  Subject: [asterisk-users] Global h exten


  Hi all,

Is there something like a global h exten, that gets called on every hang 
up, no matter what exten?

  Thanks,
  Gabriel



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Re: [asterisk-users] Global h exten

[Dovid Bender]

I had a patch created for 1.4.X for this.
http://bugs.digium.com/bug_view_page.php?bug_id=14159
  - Original Message - 
  From: Gabriel Ortiz Lour 
  To: asterisk-users@lists.digium.com 
  Sent: Wednesday, March 18, 2009 8:23 PM
  Subject: [asterisk-users] Global h exten


  Hi all,

Is there something like a global h exten, that gets called on every hang 
up, no matter what exten?

  Thanks,
  Gabriel



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Re: [asterisk-users] async agi question

[Jose Arias]

Hi,
I was asked for the patch and I sent it. Does anybody have any news about
this subject?
I'm willing to try a fix for 1.4 but I'd need any guidelines to do it.
Thanks in advanced
Jose
2009/4/2 Moises Silva moises.si...@gmail.com

 Async AGI was never released for Asterisk 1.4.X, so probably the patch
 you used has a bug or something, do you still have the patch around?

 Moy

 On Thu, Apr 2, 2009 at 5:44 AM,  cyr2...@gmail.com wrote:
  Hi Henrik,
 
  I would like to do the same thing you are doing here. I want to implement
 an external queue functionality so I need to stop a play file launched
 previously with an async agi command on caller's channel, sending the call
 to agent's extension.
 
  I'm redirecting caller's channel with REDIRECT while playing is taking
 place but I'm always getting a hang up on caller's channel.
 
  I'm using:
 
  asterisk-1.4.18
  asterisk-addons-1.4.7
  async agi patch 2007-12-11 10:34:12 (the one back-ported to 1.4)
 
  Both caller and agent are using 501 and 500 extensions and the async agi
 loop is waiting on 800, for example. The caller is dialing 800 where a play
 file is commanded through and async agi stream file command by the
 application.
 
  The relevant part of extensions.conf follows:
 
  exten = _5.,1,Noop(SIP call on 'sip_sercom' a ${EXTEN});
  exten = _5.,n,Wait(1);
  exten = _5.,n,Dial(SIP/${EXTEN},${TIMEOUTDIAL},Tto);
  exten = _5.,n,Hangup();
 
  exten = _8.,1,Noop(every thing starting 8 ${EXTEN});
  exten = _8.,n,AGI(agi:async);
  exten = _8.,n,Hangup();
 
  And the redirect command the application is sending to is:
 
  Action: Redirect
  Channel: SIP/501-081f0730
  Exten: 500
  Context: sip_sercom
  Priority: 1
 
  Therefore, Henrik, could you show me your related dial plan and the
 redirect command you are sending? I wasn't able to see what I'm getting
 wrong.
 
  thanks in advanced
  Jose M Arias
 
  --
  This message was sent on behalf of cyr2...@gmail.com at
 openSubscriber.com
 
 http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/10933120.html
 
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[asterisk-users] Off-topic: SIP DTMF most supported method

[Cesc Santa]

Hi,

I know it is a bit off-topic, but I'd like to ask the community what is the
current most supported way to deal with DTMF?
I'm looking for an all-SIP system and I'm mostly interested in the end
devices support of the different methods (DTMF in-band audio, DTMF RTP
telephony events packets, SIP INFO, ...)

Thanks in advance.

Cesc
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Re: [asterisk-users] New ViciDial Call Center Suite Release: 2.0.5

[Wolfgang Pichler]

Hi,

we are using version 2.0.4 (vicidialnow distribution) now for some time
in productino - working quit nice.

Is there any upgrade instruction out there - or will a simple yum update
do the job in the feature.

PS: On the astguiclient site you have April 3, 2008 - 
Released version 2.0.5 - i think thats not correct ;-)

regards,
Wolfgang

Am Freitag, den 03.04.2009, 10:30 -0400 schrieb Matt Florell:
 Hello,
 
 We've released another update to our VICIDIAL/astGUIclient call center
 suite: 2.0.5
 
 http://astguiclient.sf.net/
 
 The call center suite client applications run on most modern web
 browsers on almost any GUI-capable operating system, and it includes
 the VICIDIAL call center suite.
 This package is free and AGPLv2.
 This package is geared towards Asterisk installations with SIP,IAX or
 Zap phones and Zaptel, IAX or SIP trunks.
 
 For this release, we have added hundreds of new features including
 Asterisk phone, trunk and DID configuration through the VICIDIAL web
 interface. We have also tested the suite on Asterisk versions through
 1.2.30.2 and 1.4.21.2.
 
 All client web-apps and administration pages are available in English,
 Spanish, Greek, German, Italian and French, with rough translations of
 Polish, Portuguese, Brazillian Portuguese, Slovak, Russian and Dutch
 for the client web-apps only.
 
 Check out the project blog for more information:
 http://astguiclient.blogspot.com
 
 Let me know what you think.
 
 Thanks,
 
 
 MATT---
 
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Re: [asterisk-users] Please Advice SIP 183 progessl

[Khaled W. Chehab]

Dear Martin 

Can you inform me how to make the patch or from where I can get it otherwise
if there is an application can generate it?
Or if its relate to chan_sip.c ,please can you tell me which function to
edit or lines to be added

Regards


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
Sent: Sunday, April 05, 2009 5:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl

Hi Khaled,

app Dial clearly is coded to ignore the 180 Ringing being passed if
you have 'm' option to Dial and you do.
Try to take the 'm' out and see if 180 Ringing is passed to the A-leg.

So if you want MOH and then when 180 Ringing comes to turn it off =
you need a patch.

Martin

2009/4/4 Khaled W. Chehab kche...@xplorium.com:
 10x Martin ,

 But B-Leg is sending 180 ringing

 Regards

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Re: [asterisk-users] Eicon Diva 2.01 PCI Passive BRI ISDN card

[Armin Schindler]

On Mon, 6 Apr 2009, Tzafrir Cohen wrote:

On Sun, Apr 05, 2009 at 11:35:18PM +0200, Puskás Zsolt wrote:

On Sunday 05 April 2009 21.28.48 Gergo Csibra wrote:

Saturday, April 4, 2009, 3:13:12 PM, Puskás wrote:

Got it working with Asterisk 1.2 installed on the same PC as Asterisk 1.4
[ in different directorys and username of course :) ] . Using isdn4linux
kernel module and Dial(Modem/ttyI0/1234567:${EXTEN})  command.


Használj MISDN-t, és ne toppostolj.

Use MISDN, and do not toppost!


I won't toppost again but you should read my first e-mail again. I got
a passive diva isdn card which are not supported:

pc:~# mISDN scan
0 mISDN compatible device(s) found:


chan_modem is deprecated. Any chance this works with chan_capi?


Yes, but you need to use an older Version of Dialogic/Eicon driver package 
where support of these old cards was still part of. The Binary drivers Diva

PCI cards as well as the Diva PRO supported full CAPI.

Armin
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Re: [asterisk-users] Using multiple 'peer' identities on one phone with 1.4

[Florian Hackenberger]

Hi Philipp!

On Sunday 05 April 2009, Philipp von Klitzing wrote:
 Take a look at these two links:
Thanks for the links! So one option is to implement domain based 
authentication, which would be quite a bit of work. Another option 
which is quite popular is using an openSER (one of the two forks) in 
front of asterisk. Do you know if there is a way to keep sip device 
state in asterisk (for queues) working when using OpenSER as a sip 
proxy?

Cheers,
Florian

-- 
DI Florian Hackenberger
flor...@hackenberger.at
www.hackenberger.at

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Re: [asterisk-users] Fwd: add a new queue strategy: SBR

[Florian Hackenberger]

Implementing support for configuration of skills using an XML file would 
require rewriting one function. Adding the skill selections as an 
option of the queue would require a few lines of code. Apart from that 
your proposal pretty much matches my implementation.

Cheers,
Florian

On Sunday 05 April 2009, nik600 wrote:
 Thanks, this is interesting.

 I'm still looking with a customer on a possible implementation of
 sbr, this is my proposal:


 Example of skill.conf

 [default]

 ;
 ; STATIC OR DYNAMIC DEFINITION
 ;
 ;skillpath=/etc/asterisk/skills.xml
 skillpath=http://x.x.x.x/skillgenerator.php

 ; STATIC DEFINITION
 [SIP/200]
 sbr_theme=,1
 sbr_theme=,1

 [SIP/201]
 sbr_theme=,1
 sbr_theme=,1


 *
 Example of XML file located in /etc/asterisk/skills.xml / or
 generated by http://x.x.x.x/skillgenerator.php

 skills
   member interface=default
   skill theme=z1/skill
   skill theme=y2/skill
   /member
   member interface=SIP/200
   skill theme=z2/skill
   skill theme=y1/skill
   /member
   member interface=SIP/300
   skill theme=y1/skill
   skill theme=x2/skill
   /member
 /skills

 *

 you can set some variables in the channel before to queue it:

 QUEUE_SBR_THEME_z
 QUEUE_SBR_THEME_y
 QUEUE_SBR_THEME_x

 you can also set in queues.conf the theme for each queue

 [queueA]
 sbr_theme=z
 sbr_theme=y
 sbr_theme=x


 On Sun, Apr 5, 2009 at 3:57 PM, Florian Hackenberger

 f.hackenber...@chello.at wrote:
  On Sunday 08 March 2009 17:11:33 nik600 wrote:
  Hi to all isn't there any plan to add the Skills Based Routing
  strategy in queues.conf?
 
  I think that it will be enough to add an int skill to the struct
  member and then order the member by skill desc.
 
  Is it enough to add this type of strategy in calc_metric in
  app_queue.c ?
 
  Hi!
 
  I have written a patch implementing skill based routing for
  asterisk 1.4.17 (can be ported to later versions quite easily). It
  works like this:
 
  You define a database table which stores the skills:
  columns: membername, skillname, skill_level
 
  You set the strategy to skill based and set a variable for each
  incoming call which specifies which skills to take into account,
  the weight of the skill and the minimum level (optional).
 
  When selecting agents to ring, asterisk picks the agents according
  to the highest value of weighted skills (skill level multiplied by
  skill weight for all skills taken into account for that particular
  call). If an agent does not satisfy the minimum, this agent does
  not ring at all. You can for example use the minimum to make sure
  only agents speaking a particular language get a call which
  requires that language.
 
  The implementation is finished and we are currently testing it.
  Unfortunately I'm quite busy at the moment and it may take about 2
  months before I can take the time to release the code. Unless
  someone hires me as a consultant to work on it.
 
  Cheers,
         Florian
 
  --
  DI Florian Hackenberger
  flor...@hackenberger.at
  www.hackenberger.at



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www.hackenberger.at

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Re: [asterisk-users] PRI problem

[Steven J. Douglas]

Thanks for the tip, Harry. I will try that when I have exhausted all 
avenue. My problem is that if I upgrade to 1.4.24 and DAHDI, I'll break 
other stuffs.

In my current set up, the PRI did work for a long period of time (7 
hours) before going into this unreliable mode (up and down). I'm getting 
the telco technicians to check on this first because I believe the 
problem comes from their side. During the few hours when it is working, 
I am able to make and receive calls. So I don't think the issue lies 
with Asterisk.

Regards,
Steve

Harry Vangberg wrote:
 I had the exact same problem and errors some time ago (search the
 archives for PRI dropping #2) using Asterisk 1.4.18, Zaptel and a
 Digium TE121. I tried all kind of things, had telco technicians come
 out and whatnot. The solution was two-folded - 1) I reinstalled my
 server, 2) I updated to Asterisk 1.4.24, replaced Zaptel with latest
 DAHDI. In the DAHDI case I even had to use latest Subversion revision
 due to some bug (but that was related to the TE121-cards I think).
 Since then I haven't had any issues at all, so consider updating
 Asterisk and Zaptel-DAHDI

 2009/3/31 Steven J. Douglas stev...@moij.biz:
   
 Hi Brandon,

 When using the current straight cable, it sometimes worked i.e. I can
 make calls from the PSTN into the asterisk. Do you still think that I
 should try a crossover cable? Thanks.

 Regards,
 Steve.

 Brandon B. wrote:
 
 Try a T1 crossover cable:

 http://www.voip-info.org/wiki/view/crossover+T1+cable

 On Tue, Mar 31, 2009 at 12:37 AM, Steven J. Douglas stev...@moij.biz
 mailto:stev...@moij.biz wrote:

 Hi guys,

 I've been trying to get my ISDN-10 line up for the past few days, but
 its been going up and down. I am using  OpenVox  D110P  card  on
 asterisk version 1.4.21. It seems to me like a cable problem. I tried
 using Ethernet straight cable (12, 45, 36, 78) and also a straight
 cable where the twisted pairs are on 12, 34, 56 and 78. The problem
 remains the same.

 /*etc/zaptel.conf*
 loadzone=sg
 defaultzone=sg

 # PRI Span
 span=1,1,0,ccs,hdb3,crc4
 bchan=1-15
 dchan=16
 bchan=17-31


 */etc/asterisk/zapata.conf*
 language=en
 progzone=sg
 musiconhold=default

 ; PRI Set Up
 context=inbound-pri1
 switchtype=euroisdn
 signalling=pri_cpe
 pridialplan=national
 overlapdial=yes
 immediate=no
 faxdetect=both
 overlapdial=no
 usecallerid=yes
 usecallingpres=yes
 callerid=asreceived
 group=9
 channel = 1-15
 channel = 17-31


 The following are the messages that keep repeating.

  == Primary D-Channel on span 1 down
 Mar 31 14:34:05 WARNING[2361]: chan_zap.c:2682 pri_find_dchan: No
 D-channels available!  Using Primary channel 16 as D-channel anyway!
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 1
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 2
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 3
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 4
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 5
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 6
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 7
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 8
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 9
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 10
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 11
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 12
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 13
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 14
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 15
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 17
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 18
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 19
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 20
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 21
 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm
 cleared on channel 

Re: [asterisk-users] Fwd: add a new queue strategy: SBR

[Andrey Solovjov]

Mark Michelson wrote:
 Caution: One shortcoming of queue member penalties is that they are not 
 taken into account if a queue member of a low penalty does not answer a 
 call. Say for instance that the queue application determines that there 
 are two members available to answer an incoming call. One member has 
 penalty 1 and the other has penalty 2. If the member with penalty 1 does 
 not answer the call, the queue application still considers that member 
 to be available the next time that it tries to reach a member. The 
 member with penalty 2 will only be tried if the queue application can 
 determine *before the call is placed* that the member with penalty 1 is 
 unavailable.

 Mark Michelson

   
There is a patch http://bugs.digium.com/view.php?id=9165 which creates 
new strategy xrrmemory. It tries to call the member with higher penalty 
if  queue member of a lower penalty does not answer the call. 
Unfortunately disclaimer was not submitted and this patch was suspended. 
We use this patch successfully in production and in fact this is the 
most popular strategy for many of our clients. It allows to create one 
queue and you can manage which members to call first, second and so on. 
For example, you have
SIP/101,0
SIP/201,1
SIP/202,1
SIP/301,2
First queue will try to call SIP/101, then SIP/201, then SIP/202 (or 
SIP/202, SIP/201), and then SIP/301, after that it will call again 
SIP/101 and so on. You can also create simple linear strategy with 
dynamic members and they will be called in the order YOU set and not the 
order they are added to queue (current linear strategy).
I know this can be done using QUEUE_MAX_PENALTY variable but you will 
need no enter queue and leave queue many times.
This patch was written when the creator worked for our company so I can 
resubmit the code with license if it's interesting.

Andrew

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[asterisk-users] fail to retrieve the calling party information

[Rilawich Ango]

HI,

  Recently, I found that asterisk fail to get the correct context of
the sip phone.  Below is the configuration and the log message.  In
the log message, asterisk fail to identify the calling party.  As a
result, it use a default context instead of int.  Anyone know why and
how to fix it?

(testing environment)
asterisk 1.4.22  1.4.24
asterisk-addon-1.4.7

Setting
name=123
context=int

[Apr  6 17:53:12] NOTICE[4913] chan_sip.c: Call from '' to extension
'5544' rejected because extension not found.

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Re: [asterisk-users] New ViciDial Call Center Suite Release: 2.0.5

[Matt Florell]

On 4/6/09, Wolfgang Pichler wpich...@yosd.at wrote:
 Hi,

  we are using version 2.0.4 (vicidialnow distribution) now for some time
  in productino - working quit nice.

  Is there any upgrade instruction out there - or will a simple yum update
  do the job in the feature.

  PS: On the astguiclient site you have April 3, 2008 -
  Released version 2.0.5 - i think thats not correct ;-)

Hello,

There is actually an UPGRADE file right in the main directory of the
release that you should read over. Since there are many database and
dialplan changes since 2.0.4 a software-only upgrade would only get
you part of the way.

Thanks for the catch on the date, it has been fixed now.

MATT---


  Am Freitag, den 03.04.2009, 10:30 -0400 schrieb Matt Florell:

  Hello,
  
   We've released another update to our VICIDIAL/astGUIclient call center
   suite: 2.0.5
  
   http://astguiclient.sf.net/
  
   The call center suite client applications run on most modern web
   browsers on almost any GUI-capable operating system, and it includes
   the VICIDIAL call center suite.
   This package is free and AGPLv2.
   This package is geared towards Asterisk installations with SIP,IAX or
   Zap phones and Zaptel, IAX or SIP trunks.
  
   For this release, we have added hundreds of new features including
   Asterisk phone, trunk and DID configuration through the VICIDIAL web
   interface. We have also tested the suite on Asterisk versions through
   1.2.30.2 and 1.4.21.2.
  
   All client web-apps and administration pages are available in English,
   Spanish, Greek, German, Italian and French, with rough translations of
   Polish, Portuguese, Brazillian Portuguese, Slovak, Russian and Dutch
   for the client web-apps only.
  
   Check out the project blog for more information:
   http://astguiclient.blogspot.com
  
   Let me know what you think.
  
   Thanks,
  
  
   MATT---
  

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[asterisk-users] app_queue.c: No one is answering queue

[samuel]

Hi all,

Lastly we are getting several of the following errors:

app_queue.c: No one is answering queue

And when you isse a queue show XXX the status of the peers are reported as
Invalid.

We tried 1.4.23.1 and reverted back to 1.4.18.1 because it has showed good
behaviour in the past but no luck, the queue randomly stops working due to
the previous error.

We have call-limit for each agent, which are static ones.

We have checked that asterisk loads chan_sip before app_queue.

Doing reload app_queue solves the issue and the agents reports the right
status

Can anyone explain under which circumstances the

sip_devicestate
http://www.asterisk.org/doxygen/1.4/chan__sip_8c.html#bd50965f28e901cb91713b8930eaa206

functions can return Invalid??

Is there any other place to look for the possible error? network
connectivity? DNS error?

Many thanks in advance,
Samuel.
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[asterisk-users] Mysql cache delay

[cedric.bonnet]

Hi all,

I use a mysql table for sip users and I fixed rtcachefriends param to yes 
in order to have a caching of this table.

I would like to know how often does Asterisk check the mysql table to update 
its caching please.


Regards,

Cédric.

--
Cédric Bonnet
/FT/NCPI/DPS/CTR/CPM/VASF
Tel. +33 (0) 1 55 88 36 60
cedric.bon...@orange-ftgroup.com



*
This message and any attachments (the message) are confidential and intended 
solely for the addressees. 
Any unauthorised use or dissemination is prohibited.
Messages are susceptible to alteration. 
France Telecom Group shall not be liable for the message if altered, changed or 
falsified.
If you are not the intended addressee of this message, please cancel it 
immediately and inform the sender.


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Re: [asterisk-users] Asterisk Security

[SIP]

If that someone is between you and the other endpoint (like between you
and the switch, or using port-mirroring on a router somewhere), then
yes. The conversations can be recorded. In the US, the ability to be
able to do this is required by law. You've little to worry about random
hackers coming in off the Internet for this sort of thing. It's usually
something to do with having physical access to the network in which you
or the other end is connected.

There's ARP poisoning and the like which could make this possible in a
local network environment on either side, but for the most part, you'll
know who's on your local net, and they likely have physical access to
your phones as well. A listening device would be easier to plant in the
mic pickup of your phone if they REALLY wanted to listen in on your calls.

There are all sorts of levels one can to to find out what you're doing,
and preventing against them can involve a great deal of creativity.

That said, the answer is yes. You could use a VPN tunnel from one end to
the other, and many people do just that to help ensure the privacy of
their connections (both data and voice).

N.

Tom wrote:
 Since we are talking about security, if I am using * to talk to a cisco
 gateway via SIP, is there some sort of encryption you can use?  Like a 
 vpn tunnel?  

 Can someone capture packets and re-assemble to make out a conversation?



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
 Sent: Saturday, April 04, 2009 7:20 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Asterisk Security

 Lets not be that paranoid. If you have these ports open to the internet then
 from time to time someone will check if your default unsecured context
 can dial out to PSTN...

 with sip.conf you can add

 allowguest=no

 With IAX2 there's no allowguest but I believe you have to have a guest
 username in iax.conf with no password to access
 the unsecured context.

 Martin

 On Sat, Apr 4, 2009 at 3:42 PM, Todd Reese trees...@gmail.com wrote:
   
 Hi All,

 Coming in to day, the logs on the asterisk server showed several entries
 such as:

 [Apr  4 15:25:16] NOTICE[9280]: chan_sip.c:14627 handle_request_invite:
 Call from '' to extension '9810380487965419' rejected because extension
 not found.

 This has gotten me to thinking about security of this box.

 1. Currently the box sits behind a firewall with iax and sip ports
 pointing to it for the ip phones that are offsite.  There isn't any
 other access through the firewall to this box.
 2. All devices have an extension assigned to them in sip.conf and
 extensions.conf.  i.e. supra ATA, Grandstream GXP-2000
 3. The box is fed via Les.net and Voicepluse.  All other feeds are
 shutoff when not active.

 I'm looking for ideas to tighten up on the security so that this won't
 happen again.

 TIA,

 Todd Reese








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 No virus found in this incoming message.
 Checked by AVG - www.avg.com 
 Version: 8.0.238 / Virus Database: 270.11.41/2040 - Release Date: 04/04/09
 16:53:00


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Re: [asterisk-users] ISDN Timer T309

[Afonso Zimmermann]

Martin escreveu:

  What is the specification for T309 ? I'm too lazy to look it up.

The default behaviour when the alarm of layer 1 (electrical T1/E1) is
detected is to assume
all calls dropped on both sides and that's what Asterisk does.

The timer is simply deactivated since all the calls are supposed to
drop. I believe that agrees with Q921/Q931 specs.

Martin

On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann afo...@disc-os.org wrote:
  
  
Hi everione,

I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri
1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the
timer fail with a telco link in this scenario:

Telco Phone -- Telco --- Asterisk  Sip
Phone

When i make a call from Telco Phone to Sip Phone, the call complete, but
when i disconnect the link and reconnect in few seconds, the Asterisk clear
call:

[Apr 3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 1: Red Alarm
[Apr 3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got
event: Alarm (4) on Primary D-channel of span 1
 == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on
'DAHDI/1-1'
[Apr 3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No
D-channels available! Using Primary channel 16 as D-channel anyway!
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 2: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 2: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 3: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 3: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 4: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 4: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 5: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 5: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 6: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 6: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 7: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 7: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 8: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 8: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 9: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 9: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 10: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 10: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 11: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 11: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 12: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 12: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 13: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 13: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 14: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 14: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 15: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 15: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 17: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 17: Invalid argument
[Apr 3 10:44:40] 

[asterisk-users] 25-50-100fxs

[ContactTel Business]

Any hardware that can do 25-50-100 fxs ports trunked to sip ?

 

Example one end a cat5 other end 50 RJ11's jacks..

 

 

 

 

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[asterisk-users] SIP Registration and INVITE question

[Steve Davies]

I have an ITSP we are trying to work with that has an Unusual way of
working, but that said my understanding of their behaviour is that it
is fully RFC compliant. Can someone suggest how I might be able to
interoperate under these circumstances:

We register fine with them, and send the default asterisk Contact: header of:
 Contact: sip:s...@x.x.x.x

This then causes ALL calls from the ITSP inbound to us to be addressed:

 INVITE sip:s...@x.x.x.x:5060;transport=udp SIP/2.0
 To: sip:44123456...@x.x.x.x:5060;transport=udp
 [other headers omitted]

In fact, whatever we send in the Contact: header is reflected in the
INVITE for inbound calls, and the actual number dialled is always
placed in the To: header. What 99.9% of our ITSPs would send is:

 INVITE sip:44123456...@x.x.x.x:5060;transport=udp SIP/2.0
 To: sip:44123456...@x.x.x.x:5060;transport=udp
 [other headers omitted]

As you can see, the correct destination number is placed into the
INVITE header AND the To: header, and Asterisk routes it correctly
based on the INVITE.

My questions:

- Is there a way of telling chan_sip to register with multiple
Contact: headers in the registration request, so that all of the
acceptable DDI numbers can be presented to the ITSP (This is what the
RFC seems to suggest is the correct way to operate.)

- Alternatively, has anyone encountered this previously, and perhaps
created an s extension that then digs into the To: header, and
routes according to that? Examples, workarounds and solutions are all
welcome!

Help?

Thanks,
Steve

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Re: [asterisk-users] Mountain ahead of me!

[Jean-Michel Hiver]

Hello,

I want to set up a Voip Farm (c) (tm) (patent pending) but don't know
how to do it.

Please help.

Oh, the irony :)

Cheers
Jean-Michel.

2009/4/2 Gabriel - IP Guys gabr...@impactteachers.com:
 Dear All,

 Thanks for taking the time to read this. I have been presented with a massive 
 task. I'm not an asterisk expert, but I do know my way around a linux server 
 and infrastructure, and I know when things are not done correctly. A large 
 number of minutes are routed every month, (1m+) and I wish to do this in the 
 most efficient way possible.

 I've been presented with three linux servers, all in varying states of 
 upkeep, and I've decided, instead of attempting to clean the systems I'm 
 presented with, it is better for me to build a stable platform for asterisk 
 to be migrated onto. This makes my question two fold.

 1       What steps should I take, or consider, if I wish to migrate an 
 existing asterisk installation, without it being offline for too long

 2       What steps should I look out for, if I wish to move to a MySQL backed 
 for the configuration files, so that I can remove the systems dependence on 
 local configuration.

 My long term plan is to introduce MySQL to be the backend for the 
 configuration and call log data and put this machine behind a load balancer, 
 so that in due course, when I need to add more machines to handle the load, I 
 will have no need to reconfigure asterisk, or build new configurations, and 
 if I keep the base OS install uniform, I should in theory be able to deploy 
 more asterisk boxes very fast behind a load balancer to increase the capacity 
 of my VoIP Farm with minimal work.

 *VoIP farm is my term, please do not use it in any presentations to the 
 powers that be inside your organisation - If you wish to do so please send 
 £10(ten) via paypal to my email address which is clearly displayed in the 
 email headers!*

 Also, in theory, it allows for testing of new configuration, without having 
 to change the configuration on multiple machines at the same time. Which is 
 always a good thing. Any help an advice, or questions are most welcome, as I 
 wish to turn this mountain into a mole hill, a very stable, and expandable 
 mole hill!

 Thank you for your time,
 Mr Gabriel

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-- 
Jean-Michel Hiver - Synapse co-founder  CTO
GSM +262 692 828 070

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[asterisk-users] IPkall

[Dean Collins]

Does IPKALL still exist?

 

I am after a free SIP trunk - who is still giving these away these days?
As I noticed Stanaphone is no longer in business?

 

 

 

Regards,

Dean Collins
Cognation Inc
d...@cognation.net
mailto:d...@cognation.net +1-212-203-4357   New York
+61-2-9016-5642   (Sydney in-dial).
+44-20-3129-6001 (London in-dial).

 

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Re: [asterisk-users] IPkall

[SIP]

IPKall still exists.

http://www.ipkall.com

No customer service, and the number has to be used every month or you
lose it. But it's there. And free. And good.

N.

Dean Collins wrote:

 Does IPKALL still exist?

 I am after a free SIP trunk – who is still giving these away these
 days? As I noticed Stanaphone is no longer in business?

 Regards,

 Dean Collins
 Cognation Inc
 d...@cognation.net
 mailto:d...@cognation.net+1-212-203-4357 New York
 +61-2-9016-5642 (Sydney in-dial).
 +44-20-3129-6001 (London in-dial).

 

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Re: [asterisk-users] IPkall

[Daniel Nowacki]

SIP wrote:
 IPKall still exists.
 
 http://www.ipkall.com
 
 No customer service, and the number has to be used every month or you
 lose it. But it's there. And free. And good.

I get an ugly 404 when trying to sign up or log in... That is probably 
abandonware... :(

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Re: [asterisk-users] IPkall

[Anthony Francis]

SIP wrote:
 IPKall still exists.

 http://www.ipkall.com

 No customer service, and the number has to be used every month or you
 lose it. But it's there. And free. And good.

 N.

 Dean Collins wrote:
   
 Does IPKALL still exist?

 I am after a free SIP trunk – who is still giving these away these
 days? As I noticed Stanaphone is no longer in business?

 Regards,

 Dean Collins
 Cognation Inc
 d...@cognation.net
 mailto:d...@cognation.net+1-212-203-4357 New York
 +61-2-9016-5642 (Sydney in-dial).
 +44-20-3129-6001 (London in-dial).

 
The sign up link doesn't work.

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Re: [asterisk-users] IPkall

[Dean Collins]

None of their pages apart from the front page seem to work though
http://phone.ipkall.com/ipphone/login.asp


Are you sure they still exist?

 

 

Regards,

Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357   New York
+61-2-9016-5642   (Sydney in-dial).
+44-20-3129-6001 (London in-dial).


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of SIP
Sent: Monday, April 06, 2009 9:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IPkall

IPKall still exists.

http://www.ipkall.com

No customer service, and the number has to be used every month or you
lose it. But it's there. And free. And good.

N.

Dean Collins wrote:

 Does IPKALL still exist?

 I am after a free SIP trunk - who is still giving these away these
 days? As I noticed Stanaphone is no longer in business?

 Regards,

 Dean Collins
 Cognation Inc
 d...@cognation.net
 mailto:d...@cognation.net+1-212-203-4357 New York
 +61-2-9016-5642 (Sydney in-dial).
 +44-20-3129-6001 (London in-dial).




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[asterisk-users] OT - Call forwarding services for corporate users

[Olivier]

Hello,

For corporate users, how would you define Call Forwarding services ?

1. Would offer option A or B ?
option A:
no forwarding
immediate
busy
no answer

option B:
no forwarding
immediate
busy
no answer
busy or no answer

I've seen legacy PBX offering B and SIP phones offering A.
Which is the trend ?


2. If an extension is unconditionally forwarded (immediate forwarding), and
destination is either busy, not answering, ... it's possible to leave
voicemail into
original or forwarded extension, or leave no message at all.
Which option would be more natural ?
C: leave voicemail into original extension's voicemail
D: leave voicemail into forwarded extension's voicemail
E: leave no voicemail at all
F: have it user configurable


Regards
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Re: [asterisk-users] 25-50-100fxs

[Jeff LaCoursiere]

On Mon, 6 Apr 2009, ContactTel Business wrote:

 Any hardware that can do 25-50-100 fxs ports trunked to sip ?

 Example one end a cat5 other end 50 RJ11's jacks..


Audiocodes MP-124 has 24 FXS ports and an amphenol connector (so you would 
need to add a breakout box to get your RJ11 jacks).

j

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Re: [asterisk-users] IPkall

[Steve Howes]

On 6 Apr 2009, at 14:32, Dean Collins wrote:

 None of their pages apart from the front page seem to work though
 http://phone.ipkall.com/ipphone/login.asp

http://phone.ipkall.com/login.asp

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Re: [asterisk-users] 25-50-100fxs

[ContactTel Business]

Hmm, this seem to be the biggest non cisco device i found as well, 
The breakout is a FXS, 50-pin Telco to rj11 converter ?



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: April-06-09 9:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 25-50-100fxs


On Mon, 6 Apr 2009, ContactTel Business wrote:

 Any hardware that can do 25-50-100 fxs ports trunked to sip ?

 Example one end a cat5 other end 50 RJ11's jacks..


Audiocodes MP-124 has 24 FXS ports and an amphenol connector (so you would 
need to add a breakout box to get your RJ11 jacks).

j

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Re: [asterisk-users] 25-50-100fxs

[Jeff LaCoursiere]

On Mon, 6 Apr 2009, ContactTel Business wrote:

 Hmm, this seem to be the biggest non cisco device i found as well,
 The breakout is a FXS, 50-pin Telco to rj11 converter ?


Yes.  MP-124 is a solid, stable device.  Any vendor that sells you the 
MP-124 will have a breakout box (or patch panel) to sell you to get you 
your RJ-11 jacks.

j



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
 LaCoursiere
 Sent: April-06-09 9:35 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] 25-50-100fxs


 On Mon, 6 Apr 2009, ContactTel Business wrote:

 Any hardware that can do 25-50-100 fxs ports trunked to sip ?

 Example one end a cat5 other end 50 RJ11's jacks..


 Audiocodes MP-124 has 24 FXS ports and an amphenol connector (so you would
 need to add a breakout box to get your RJ11 jacks).

 j

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Re: [asterisk-users] 25-50-100fxs

[ContactTel Business]

Grandstream GXW4024 IP Analog Gateway

Also seem to do it, not sure what is better between AC and GS..
I think AC is more complicated to program but better quality, while GS is
half the price of the AC.

But also comes with rj11 jacks.. the AC has a 50 pin

Any opinions ?

Basically need to wire 10 hotels ;)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: April-06-09 9:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 25-50-100fxs


On Mon, 6 Apr 2009, ContactTel Business wrote:

 Any hardware that can do 25-50-100 fxs ports trunked to sip ?

 Example one end a cat5 other end 50 RJ11's jacks..


Audiocodes MP-124 has 24 FXS ports and an amphenol connector (so you would 
need to add a breakout box to get your RJ11 jacks).

j

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Re: [asterisk-users] IPkall

[Jaswinder Singh]

I registered few days back and got a DID. Maybe this is temporary ?

On Mon, Apr 6, 2009 at 7:05 PM, Steve Howes st...@geekinter.net wrote:

 On 6 Apr 2009, at 14:32, Dean Collins wrote:

  None of their pages apart from the front page seem to work though
  http://phone.ipkall.com/ipphone/login.asp

 http://phone.ipkall.com/login.asp

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Re: [asterisk-users] 25-50-100fxs

[Jeff LaCoursiere]

On Mon, 6 Apr 2009, ContactTel Business wrote:

 Grandstream GXW4024 IP Analog Gateway

 Also seem to do it, not sure what is better between AC and GS..
 I think AC is more complicated to program but better quality, while GS is
 half the price of the AC.

 But also comes with rj11 jacks.. the AC has a 50 pin

 Any opinions ?

 Basically need to wire 10 hotels ;)

Depends how often you want to visit those hotels to support their 
equipment.  I've had nothing but problems with Grandstream.  I've left 
Audiocodes boxes at installations and not touched them for four years.  My 
Grandstream experience is several years old at this point, however. 
Perhaps they are more stable now.  I was constantly rebooting the few 8 
port boxes I tried to use back then, and voice quality was not good.

j


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
 LaCoursiere
 Sent: April-06-09 9:35 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] 25-50-100fxs


 On Mon, 6 Apr 2009, ContactTel Business wrote:

 Any hardware that can do 25-50-100 fxs ports trunked to sip ?

 Example one end a cat5 other end 50 RJ11's jacks..


 Audiocodes MP-124 has 24 FXS ports and an amphenol connector (so you would
 need to add a breakout box to get your RJ11 jacks).

 j

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Re: [asterisk-users] 25-50-100fxs

[ContactTel Business]

Actually thinking about it, that 50 pins is simply the 48 + 2 grounds i
imagine.. or something of the likes.. 
Thanks Jeff, you have pointed me in the right direction.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: April-06-09 9:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 25-50-100fxs


On Mon, 6 Apr 2009, ContactTel Business wrote:

 Hmm, this seem to be the biggest non cisco device i found as well,
 The breakout is a FXS, 50-pin Telco to rj11 converter ?


Yes.  MP-124 is a solid, stable device.  Any vendor that sells you the 
MP-124 will have a breakout box (or patch panel) to sell you to get you 
your RJ-11 jacks.

j



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
 LaCoursiere
 Sent: April-06-09 9:35 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] 25-50-100fxs


 On Mon, 6 Apr 2009, ContactTel Business wrote:

 Any hardware that can do 25-50-100 fxs ports trunked to sip ?

 Example one end a cat5 other end 50 RJ11's jacks..


 Audiocodes MP-124 has 24 FXS ports and an amphenol connector (so you would
 need to add a breakout box to get your RJ11 jacks).

 j

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Re: [asterisk-users] IPkall

[SIP]

Daniel Nowacki wrote:
 SIP wrote:
   
 IPKall still exists.

 http://www.ipkall.com

 No customer service, and the number has to be used every month or you
 lose it. But it's there. And free. And good.
 

 I get an ugly 404 when trying to sign up or log in... That is probably 
 abandonware... :(

   
No. It's just poorly-checked web management.

http://phone.ipkall.com/

Is the signup link.  The /ipphone stuff appears to be an old document
tree that no longer exists.

N.

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[asterisk-users] Douds it

[jibanez1971]

I have a few questions.

 

Asterisk is a windows program why each time I try to find out how
communicate with my Panasonic TDA 100 or with TDE 100 always read use
one card o use a box why I can't use simply my network card, in the
other side of Panasonic exist two types of cards one in TDA 100 with 2
trunks and in the other side TDE have internal Two trunks too. Why if I
want to use asterisk in home and I have a modem with voice I can't use
it to access to a line. ?

 

Apologize my English

Regards

 

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Re: [asterisk-users] Douds it

[zoach...@securax.org]

jibanez1...@cimex.com.cu wrote:

 I have a few questions.

 Asterisk is a windows program

Asterisk is not a windows program.

 why each time I try to find out how communicate with my Panasonic TDA 
 100 or with TDE 100 always read “use one card o use a box” why I can’t 
 use simply my network card, in the other side of Panasonic exist two 
 types of cards one in TDA 100 with 2 trunks and in the other side TDE 
 have internal Two trunks too.

If your panasonic allows SIP or h323 that might work.
A quick google search shows the ethernet port is for configuration of 
the IVR only, so i guess that is a no.
It does do TAPI,so if a tapi plugin for asterisk exists, this might work.

 Why if I want to use asterisk in home and I have a modem with voice I 
 can’t use it to access to a line. ?

Because you will need a driver for that card for asterisk.


 Apologize my English

 Regards

 

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Re: [asterisk-users] 25-50-100fxs

[Luis Morales]

Take a look on xorcom solutions

http://www.youtube.com/watch?v=qt4aPdGIvIQfeature=player_embedded

Regards,

Luis Morales

On Mon, Apr 6, 2009 at 8:09 AM, ContactTel Business
li...@contacttel.com wrote:
 Any hardware that can do 25-50-100 fxs ports trunked to sip ?



 Example one end a cat5 other end 50 RJ11’s jacks..









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-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible

Leonardo Da'Vinci
-

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Re: [asterisk-users] async agi question

[Moises Silva]

You have to understand that this mailing list is not free instant
support. Even more, you are using an unsupported Asterisk feature for
1.4. I will check it when I have some spare time to try to reproduce
and fix it. If you are too much in a hurry you can always contact me
off-list for paid support for this feature.

Moy

On Mon, Apr 6, 2009 at 3:15 AM, Jose Arias cyr2...@gmail.com wrote:
 Hi,
 I was asked for the patch and I sent it. Does anybody have any news about
 this subject?
 I'm willing to try a fix for 1.4 but I'd need any guidelines to do it.
 Thanks in advanced
 Jose
 2009/4/2 Moises Silva moises.si...@gmail.com

 Async AGI was never released for Asterisk 1.4.X, so probably the patch
 you used has a bug or something, do you still have the patch around?

 Moy

 On Thu, Apr 2, 2009 at 5:44 AM,  cyr2...@gmail.com wrote:
  Hi Henrik,
 
  I would like to do the same thing you are doing here. I want to
  implement an external queue functionality so I need to stop a play file
  launched previously with an async agi command on caller's channel, sending
  the call to agent's extension.
 
  I'm redirecting caller's channel with REDIRECT while playing is taking
  place but I'm always getting a hang up on caller's channel.
 
  I'm using:
 
  asterisk-1.4.18
  asterisk-addons-1.4.7
  async agi patch 2007-12-11 10:34:12 (the one back-ported to 1.4)
 
  Both caller and agent are using 501 and 500 extensions and the async agi
  loop is waiting on 800, for example. The caller is dialing 800 where a play
  file is commanded through and async agi stream file command by the
  application.
 
  The relevant part of extensions.conf follows:
 
  exten = _5.,1,Noop(SIP call on 'sip_sercom' a ${EXTEN});
  exten = _5.,n,Wait(1);
  exten = _5.,n,Dial(SIP/${EXTEN},${TIMEOUTDIAL},Tto);
  exten = _5.,n,Hangup();
 
  exten = _8.,1,Noop(every thing starting 8 ${EXTEN});
  exten = _8.,n,AGI(agi:async);
  exten = _8.,n,Hangup();
 
  And the redirect command the application is sending to is:
 
  Action: Redirect
  Channel: SIP/501-081f0730
  Exten: 500
  Context: sip_sercom
  Priority: 1
 
  Therefore, Henrik, could you show me your related dial plan and the
  redirect command you are sending? I wasn't able to see what I'm getting
  wrong.
 
  thanks in advanced
  Jose M Arias
 
  --
  This message was sent on behalf of cyr2...@gmail.com at
  openSubscriber.com
 
  http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/10933120.html
 
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 --
 I do not agree with what you have to say, but I’ll defend to the
 death your right to say it. Voltaire


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-- 
I do not agree with what you have to say, but I’ll defend to the
death your right to say it. Voltaire

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[asterisk-users] Relay ringing sip message 180

[Khaled W. Chehab]

Dears

 

Asterisk is a median server between the caller and the terminations gateway

 

The  caller send the call to asterisk à asterisk will play music on hold
untill the termination gateway send 200 OK and the RTP establish

My problem that, Asterisk is not forwarding the 180 ringing  from the
termination gateway to the user 

How can I forward the sip message 180 to the caller or let the music on hold
stop playing  and  the caller hears  the Ring Back Tone which  when 180
ringing from the termination gateway.

What I am using now to stop the musinc on hold when the RTP established is 

exten = _X.,n,Dial(SIP/Temination_Gateway/${EXTEN}|300|m)

 

NB: I tried to edit chan_sip.c  but I did not find the solution 

 

Please Advice

 

 

Regards

 

 

 



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Re: [asterisk-users] Relay ringing sip message 180

[Steve Howes]

On 6 Apr 2009, at 15:40, Khaled W. Chehab wrote:

 Dears

 Asterisk is a median server between the caller and the terminations  
 gateway

 The  caller send the call to asterisk à asterisk will play music on  
 hold untill the termination gateway send 200 OK and the RTP establish
 My problem that, Asterisk is not forwarding the 180 ringing  from  
 the termination gateway to the user
 How can I forward the sip message 180 to the caller or let the music  
 on hold stop playing  and  the caller hears  the Ring Back Tone  
 which  when 180 ringing from the termination gateway.
 What I am using now to stop the musinc on hold when the RTP  
 established is
 exten = _X.,n,Dial(SIP/Temination_Gateway/${EXTEN}|300|m)

 NB: I tried to edit chan_sip.c  but I did not find the solution

 Please Advice

FFS stop posting the same crap over and over again, thats 4 times now
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Re: [asterisk-users] Asterisk + Cisco Call Manager

[David Backeberg]

On Sat, Apr 4, 2009 at 11:18 AM, Timothy Smith timotsm...@gmail.com wrote:
 We're migrating from Cisco to asterisk because cisco is expensive to
 maintain, besides we can achieve more with asterisk like customised
 IVRs etc.

I don't know what expensive to maintain means. We spend more on our
phone bill than what the gear costs by a significant margin.

 This being a large production environment, we can't just change over
 without testing thoroughly..

makes sense

Now, i'd like to
 completely get rid of the cisco gateways by routing incoming calls
 through asterisk too (to the call manager, and finally the phones).

I don't have an architecture diagram of your call flow, but you'll
need to be picking off calls with some criteria, perhaps the number
dialed or DNIS if these are PRIs and route accordingly. You should be
able to make asterisk talk directly to your phones by putting some
asterisk hardware onto the same address space as the phones. If these
are Cisco phones most of them are multi-line, and you can go ahead and
push a change that one of the lines will SIP-register with an asterisk
system. At that point you don't need to route _through_ Cisco because
you would be routing around Call Manager.

If you want to keep Call Manager in the loop you need to have the
traffic going to Cisco continue to look like it did before you put
Asterisk in the loop, or you need to change the Call Manager to act
according to the revised traffic it will be receiving from Asterisk. I
recommend enabling call debugging on the Asterisk side and the Cisco
side, and figure out which side(s) you want to change to keep both
sides happy.

 After that, they'll be satisfied and i'll start registering the phones
 to asterisk until everything is asterisk. It has to be a smooth
 transition, just fyi, we're about 200 employees.
 I'll appreciate any advice towards achieving this.

My advice is to test changes in a small lab setup before you cut them
loose, assuming that no downtime is acceptable. No idea what your
budget is, nor your type of call traffic. If this is real telco lines,
you may need a test telco line for your experiments. (which you can
also fake with asterisk and/or Cisco). If this is just SIP you can
fake it with asterisk and cheaper used Cisco gear. Hopefully you
already have a lab environment where you can test things out.

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[asterisk-users] [OT] Re: async agi question

[Philipp Kempgen]

cyr2...@gmail.com schrieb:
 This message was sent on behalf of cyr2...@gmail.com at openSubscriber.com

Use the appropriate header field for that information.
It's called From (in contrast to Sender).


Philipp Kempgen
-- 
AMOOCON 2009, May 4-5, Rostock / Germany   -  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
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Re: [asterisk-users] ISDN Timer T309

[Martin]

Based on the Asterisk logs you posted the Asterisk doesn't have it
implemented per:

The implementation of timer T309 in the user side is optional

Martin

On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann afo...@disc-os.org wrote:
 Martin escreveu:

 What is the specification for T309 ? I'm too lazy to look it up.

 The default behaviour when the alarm of layer 1 (electrical T1/E1) is
 detected is to assume
 all calls dropped on both sides and that's what Asterisk does.

 The timer is simply deactivated since all the calls are supposed to
 drop. I believe that agrees with Q921/Q931 specs.

 Martin

 On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann afo...@disc-os.org
 wrote:


 Hi everione,

 I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri
 1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the
 timer fail with a telco link in this scenario:

 Telco Phone -- Telco --- Asterisk  Sip
 Phone

 When i make a call from Telco Phone to Sip Phone, the call complete, but
 when i disconnect the link and reconnect in few seconds, the Asterisk clear
 call:

 [Apr  3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 1: Red Alarm
 [Apr  3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got
 event: Alarm (4) on Primary D-channel of span 1
   == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on
 'DAHDI/1-1'
 [Apr  3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No
 D-channels available!  Using Primary channel 16 as D-channel anyway!
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 2: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 2: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 3: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 3: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 4: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 4: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 5: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 5: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 6: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 6: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 7: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 7: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 8: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 8: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 9: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 9: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 10: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 10: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 11: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 11: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 12: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 12: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 13: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 13: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 14: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 14: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 15: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: 

Re: [asterisk-users] Please Advice SIP 183 progessl

[Martin]

Hi,

The easiest is to turn off MOH on the Dial. Otherwise the patch is
easy but not trivial.
Once the B-leg receives the ringing message and passes it in Dial app
then the code has to turn off the MOH
and tell the A-leg to send the ringing message. At the same time the
code that skips passing the ringing to A-leg
has to be disabled.

Martin

On Mon, Apr 6, 2009 at 2:38 AM, Khaled W. Chehab kche...@xplorium.com wrote:
 Dear Martin

 Can you inform me how to make the patch or from where I can get it otherwise
 if there is an application can generate it?
 Or if its relate to chan_sip.c ,please can you tell me which function to
 edit or lines to be added

 Regards


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
 Sent: Sunday, April 05, 2009 5:06 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Please Advice SIP 183 progessl

 Hi Khaled,

 app Dial clearly is coded to ignore the 180 Ringing being passed if
 you have 'm' option to Dial and you do.
 Try to take the 'm' out and see if 180 Ringing is passed to the A-leg.

 So if you want MOH and then when 180 Ringing comes to turn it off =
 you need a patch.

 Martin

 2009/4/4 Khaled W. Chehab kche...@xplorium.com:
 10x Martin ,

 But B-Leg is sending 180 ringing

 Regards

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[asterisk-users] IOS Interface

[Jorge Mendoza]

Are there an IOS interface for Asterisk?, or an IOS to SIP converter?
Some femtocells uses this protocol and I would to use them with Asterisk.

Jorge Mendoza

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Re: [asterisk-users] Off-topic: SIP DTMF most supported method

[Martin]

It's SIP in rfc (RFC2833) then SIP INFO and then if you can't do
anything else inband audio (only G711)

Martin

On Mon, Apr 6, 2009 at 2:24 AM, Cesc Santa cesc.sa...@gmail.com wrote:
 Hi,

 I know it is a bit off-topic, but I'd like to ask the community what is the
 current most supported way to deal with DTMF?
 I'm looking for an all-SIP system and I'm mostly interested in the end
 devices support of the different methods (DTMF in-band audio, DTMF RTP
 telephony events packets, SIP INFO, ...)

 Thanks in advance.
 Cesc
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Re: [asterisk-users] SIP Registration and INVITE question

[Martin]

Have you looked at the syntax of register = keyword ?

register = [transport://]user[:secret[:authuse...@host[:port][/extension]
; If no extension is given, the 's' extension is used.

There you have it ... Contact: sip:s 

set the extension and you should be fine

Martin

On Mon, Apr 6, 2009 at 7:45 AM, Steve Davies davies...@gmail.com wrote:
 I have an ITSP we are trying to work with that has an Unusual way of
 working, but that said my understanding of their behaviour is that it
 is fully RFC compliant. Can someone suggest how I might be able to
 interoperate under these circumstances:

 We register fine with them, and send the default asterisk Contact: header of:
     Contact: sip:s...@x.x.x.x

 This then causes ALL calls from the ITSP inbound to us to be addressed:

     INVITE sip:s...@x.x.x.x:5060;transport=udp SIP/2.0
     To: sip:44123456...@x.x.x.x:5060;transport=udp
     [other headers omitted]

 In fact, whatever we send in the Contact: header is reflected in the
 INVITE for inbound calls, and the actual number dialled is always
 placed in the To: header. What 99.9% of our ITSPs would send is:

     INVITE sip:44123456...@x.x.x.x:5060;transport=udp SIP/2.0
     To: sip:44123456...@x.x.x.x:5060;transport=udp
     [other headers omitted]

 As you can see, the correct destination number is placed into the
 INVITE header AND the To: header, and Asterisk routes it correctly
 based on the INVITE.

 My questions:

 - Is there a way of telling chan_sip to register with multiple
 Contact: headers in the registration request, so that all of the
 acceptable DDI numbers can be presented to the ITSP (This is what the
 RFC seems to suggest is the correct way to operate.)

 - Alternatively, has anyone encountered this previously, and perhaps
 created an s extension that then digs into the To: header, and
 routes according to that? Examples, workarounds and solutions are all
 welcome!

 Help?

 Thanks,
 Steve

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Re: [asterisk-users] fail to retrieve the calling party information

[Martin]

That's because you have to create a user account in sip.conf ... +
Asterisk is sensitive about it.
What should help is if you register the phone with that sip account first.

Martin

On Mon, Apr 6, 2009 at 5:00 AM, Rilawich Ango maillist...@gmail.com wrote:
 HI,

  Recently, I found that asterisk fail to get the correct context of
 the sip phone.  Below is the configuration and the log message.  In
 the log message, asterisk fail to identify the calling party.  As a
 result, it use a default context instead of int.  Anyone know why and
 how to fix it?

 (testing environment)
 asterisk 1.4.22  1.4.24
 asterisk-addon-1.4.7

 Setting
 name=123
 context=int

 [Apr  6 17:53:12] NOTICE[4913] chan_sip.c: Call from '' to extension
 '5544' rejected because extension not found.

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Re: [asterisk-users] SIP Registration and INVITE question

[Steve Davies]

Thanks for the reply - Perhaps I was not clear.

On the register= line, if I set /extension to be /12345, then this
just replaces 's' with 12345, and ALL calls, regardless of their
destination number will be routed on the INVITE line to 12...@x.x.x.x,
and the actual destination is specified in the To: header.

Not particularly useful, and I'd prefer not to have to go fumbling
through the SIP headers to find what was really dialled :)

Looking at the SIP RFC, the idea is that you specify a set of What I
will accept details with each registration in the Contact: headers,
which is intended to include _multiple_ possible destination
addresses. The Registrar will then only ever send calls addressed to
that list of destinations. Sadly, the RFC authors did not think to
consider private point-to-point links where you can usefully say send
me anything, you know best. Asterisk fills by defaulting to a
single s...@x.x.x.x, where the 's' can be replaced by any single number.

Most ITSPs work around this by assuming that they know best, and
routing numbers even if they are missing from the registration. The
odd exception does not do this.

I suspect that the solution will be to register with a /extension of
/pedanticitsp, and then have a dialplan which pulls and parses the SIP
To: header. Other suggestions are still welcome.

Regards,
Steve

2009/4/6 Martin asteriskl...@callthem.info:
 Have you looked at the syntax of register = keyword ?

 register = [transport://]user[:secret[:authuse...@host[:port][/extension]
 ; If no extension is given, the 's' extension is used.

 There you have it ... Contact: sip:s 

 set the extension and you should be fine

 Martin

 On Mon, Apr 6, 2009 at 7:45 AM, Steve Davies davies...@gmail.com wrote:
 I have an ITSP we are trying to work with that has an Unusual way of
 working, but that said my understanding of their behaviour is that it
 is fully RFC compliant. Can someone suggest how I might be able to
 interoperate under these circumstances:

 We register fine with them, and send the default asterisk Contact: header of:
     Contact: sip:s...@x.x.x.x

 This then causes ALL calls from the ITSP inbound to us to be addressed:

     INVITE sip:s...@x.x.x.x:5060;transport=udp SIP/2.0
     To: sip:44123456...@x.x.x.x:5060;transport=udp
     [other headers omitted]

 In fact, whatever we send in the Contact: header is reflected in the
 INVITE for inbound calls, and the actual number dialled is always
 placed in the To: header. What 99.9% of our ITSPs would send is:

     INVITE sip:44123456...@x.x.x.x:5060;transport=udp SIP/2.0
     To: sip:44123456...@x.x.x.x:5060;transport=udp
     [other headers omitted]

 As you can see, the correct destination number is placed into the
 INVITE header AND the To: header, and Asterisk routes it correctly
 based on the INVITE.

 My questions:

 - Is there a way of telling chan_sip to register with multiple
 Contact: headers in the registration request, so that all of the
 acceptable DDI numbers can be presented to the ITSP (This is what the
 RFC seems to suggest is the correct way to operate.)

 - Alternatively, has anyone encountered this previously, and perhaps
 created an s extension that then digs into the To: header, and
 routes according to that? Examples, workarounds and solutions are all
 welcome!

 Help?

 Thanks,
 Steve

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Re: [asterisk-users] 25-50-100fxs

[ContactTel Business]

Why would i want to do that ?



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales
Sent: April-06-09 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 25-50-100fxs

Take a look on xorcom solutions

http://www.youtube.com/watch?v=qt4aPdGIvIQfeature=player_embedded

Regards,

Luis Morales

On Mon, Apr 6, 2009 at 8:09 AM, ContactTel Business
li...@contacttel.com wrote:
 Any hardware that can do 25-50-100 fxs ports trunked to sip ?



 Example one end a cat5 other end 50 RJ11’s jacks..









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-- 

-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091

-
Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible

Leonardo Da'Vinci

-

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Re: [asterisk-users] IOS Interface

[Brent Davidson]

Jorge Mendoza wrote:
 Are there an IOS interface for Asterisk?, or an IOS to SIP converter?
 Some femtocells uses this protocol and I would to use them with Asterisk.

 Jorge Mendoza

 ___
   

You're comparing to apples to Orange.  IOS is the Cisco operating system 
run by the Femtocells, not the protocol.  I'm not familiar with 
Femtocells, but as far as I can tell (from reading wikipedia) they 
apparently  can do SIP internally, but that is a more advance 
configuration and might require some additional software.  Looks like 
they are more designed to what is called lub over IP which appears to 
be some sort of backhaul specification specific to Cellular / Wireless 
carrier technology.

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Re: [asterisk-users] ISDN Timer T309

[Afonso Zimmermann]

Martin escreveu:

  Based on the Asterisk logs you posted the Asterisk doesn't have it
implemented per:

"The implementation of timer T309 in the user side is optional"

Martin

On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann afo...@disc-os.org wrote:
  
  
Martin escreveu:

What is the specification for T309 ? I'm too lazy to look it up.

The default behaviour when the alarm of layer 1 (electrical T1/E1) is
detected is to assume
all calls dropped on both sides and that's what Asterisk does.

The timer is simply deactivated since all the calls are supposed to
drop. I believe that agrees with Q921/Q931 specs.

Martin

On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann afo...@disc-os.org
wrote:


Hi everione,

I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri
1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the
timer fail with a telco link in this scenario:

Telco Phone -- Telco --- Asterisk  Sip
Phone

When i make a call from Telco Phone to Sip Phone, the call complete, but
when i disconnect the link and reconnect in few seconds, the Asterisk clear
call:

[Apr  3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 1: Red Alarm
[Apr  3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got
event: Alarm (4) on Primary D-channel of span 1
  == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on
'DAHDI/1-1'
[Apr  3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No
D-channels available!  Using Primary channel 16 as D-channel anyway!
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 2: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 2: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 3: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 3: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 4: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 4: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 5: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 5: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 6: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 6: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 7: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 7: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 8: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 8: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 9: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 9: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 10: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 10: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 11: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 11: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 12: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 12: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 13: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 13: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 14: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 14: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 15: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on 

Re: [asterisk-users] 25-50-100fxs

[Luis Morales]

This may be your solution.

Regards,

Luis Morales

On Mon, Apr 6, 2009 at 12:23 PM, ContactTel Business
li...@contacttel.com wrote:
 Why would i want to do that ?



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales
 Sent: April-06-09 10:27 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] 25-50-100fxs

 Take a look on xorcom solutions

 http://www.youtube.com/watch?v=qt4aPdGIvIQfeature=player_embedded

 Regards,

 Luis Morales

 On Mon, Apr 6, 2009 at 8:09 AM, ContactTel Business
 li...@contacttel.com wrote:
 Any hardware that can do 25-50-100 fxs ports trunked to sip ?



 Example one end a cat5 other end 50 RJ11’s jacks..









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




 --
 
 -
 Luis Morales
 Consultor de Tecnologia
 Cel: +(58)416-4242091
 
 -
 Empieza por hacer lo necesario, luego lo que es posible... y de
 pronto estarás haciendo lo imposible

 Leonardo Da'Vinci
 
 -

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-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible

Leonardo Da'Vinci
-

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Re: [asterisk-users] 25-50-100fxs

[ContactTel Business]

Aint this based on asterisk ?

I don't think i would use that, thanks anyway.
And yes i know this is an asterisk list.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales
Sent: April-06-09 2:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 25-50-100fxs

This may be your solution.

Regards,

Luis Morales

On Mon, Apr 6, 2009 at 12:23 PM, ContactTel Business
li...@contacttel.com wrote:
 Why would i want to do that ?



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales
 Sent: April-06-09 10:27 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] 25-50-100fxs

 Take a look on xorcom solutions

 http://www.youtube.com/watch?v=qt4aPdGIvIQfeature=player_embedded

 Regards,

 Luis Morales

 On Mon, Apr 6, 2009 at 8:09 AM, ContactTel Business
 li...@contacttel.com wrote:
 Any hardware that can do 25-50-100 fxs ports trunked to sip ?



 Example one end a cat5 other end 50 RJ11’s jacks..









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




 --


 -
 Luis Morales
 Consultor de Tecnologia
 Cel: +(58)416-4242091


 -
 Empieza por hacer lo necesario, luego lo que es posible... y de
 pronto estarás haciendo lo imposible

 Leonardo Da'Vinci


 -

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-- 

-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091

-
Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible

Leonardo Da'Vinci

-

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Re: [asterisk-users] IOS Interface

[Brent Davidson]

Jorge Mendoza wrote:

Brent Davidson wrote:
  

Jorge Mendoza wrote:
  


Are there an IOS interface for Asterisk?, or an IOS to SIP converter?
Some femtocells uses this protocol and I would to use them with Asterisk.

Jorge Mendoza

___
  

  
You're comparing to apples to Orange.  IOS is the Cisco operating system 
run by the Femtocells, not the protocol.  I'm not familiar with 
Femtocells, but as far as I can tell (from reading wikipedia) they 
apparently  can do SIP internally, but that is a more advance 
configuration and might require some additional software.  Looks like 
they are more designed to what is called lub over IP which appears to 
be some sort of backhaul specification specific to Cellular / Wireless 
carrier technology.
  


AFAIK, IOS interface for femtocell is a 3GPP2 specification. It is an
application, no related to Cisco OS.

Jorge Mendoza

  
Interesting.  I've never seen anything refer to IOS other than in the 
context of the OS run by Cisco routers although with so many acronyms 
around I suppose it's just a given that some of them should have more 
than one meaning depending on the context.


Anyway, as I've already gone way past my level of understanding on the 
subject I'll leave this thread to someone more qualified to weigh in.  :-P



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Re: [asterisk-users] 25-50-100fxs

[ContactTel Business]

Actually i might rephrase, i need hardware solution not pc based, no hard
drives, no fans, no application you need to monitor, hence hardware, 

You can ignore the rest of this thread i have my info.

Thanks


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ContactTel
Business
Sent: April-06-09 2:19 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] 25-50-100fxs

Aint this based on asterisk ?

I don't think i would use that, thanks anyway.
And yes i know this is an asterisk list.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales
Sent: April-06-09 2:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 25-50-100fxs

This may be your solution.

Regards,

Luis Morales

On Mon, Apr 6, 2009 at 12:23 PM, ContactTel Business
li...@contacttel.com wrote:
 Why would i want to do that ?



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales
 Sent: April-06-09 10:27 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] 25-50-100fxs

 Take a look on xorcom solutions

 http://www.youtube.com/watch?v=qt4aPdGIvIQfeature=player_embedded

 Regards,

 Luis Morales

 On Mon, Apr 6, 2009 at 8:09 AM, ContactTel Business
 li...@contacttel.com wrote:
 Any hardware that can do 25-50-100 fxs ports trunked to sip ?



 Example one end a cat5 other end 50 RJ11’s jacks..









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




 --


 -
 Luis Morales
 Consultor de Tecnologia
 Cel: +(58)416-4242091


 -
 Empieza por hacer lo necesario, luego lo que es posible... y de
 pronto estarás haciendo lo imposible

 Leonardo Da'Vinci


 -

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-- 

-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091

-
Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible

Leonardo Da'Vinci

-

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Re: [asterisk-users] Provisioning GXP 2000

[Philipp Kempgen]

David Ruggles schrieb:
 I've done some googling and searched voip-info but I'm not able to find a
 good answer about how to provision the GXP 2000.
 
 Based on questions I've asked before it seems like a lot of people are using
 the grandstream phones so I figure provisioning can't be that hard. Is
 everyone using the web interface for *every* phone? Or is there a better,
 more automatic, way?

Here's our source code (GPL) for provisioning of Grandstream phones:

https://svn.amooma.com/gemeinschaft/trunk/opt/gemeinschaft/htdocs/prov/grandstream/settings.php
https://svn.amooma.com/gemeinschaft/trunk/opt/gemeinschaft/htdocs/prov/grandstream/.htaccess

Works for the BT 110, BT 200, GXP 1200, GXP 2000, GXP 2010,
GXP 2020. Not sure about the GXV 3000.


Philipp Kempgen
-- 
AMOOCON 2009, May 4-5, Rostock / Germany   -  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] IOS Interface

[Jorge Mendoza]

Brent Davidson wrote:
 Jorge Mendoza wrote:
   
 Are there an IOS interface for Asterisk?, or an IOS to SIP converter?
 Some femtocells uses this protocol and I would to use them with Asterisk.

 Jorge Mendoza

 ___
   
 

 You're comparing to apples to Orange.  IOS is the Cisco operating system 
 run by the Femtocells, not the protocol.  I'm not familiar with 
 Femtocells, but as far as I can tell (from reading wikipedia) they 
 apparently  can do SIP internally, but that is a more advance 
 configuration and might require some additional software.  Looks like 
 they are more designed to what is called lub over IP which appears to 
 be some sort of backhaul specification specific to Cellular / Wireless 
 carrier technology.
   
AFAIK, IOS interface for femtocell is a 3GPP2 specification. It is an
application, no related to Cisco OS.

Jorge Mendoza

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[asterisk-users] IMAP Voicemail - can't get messages. Arrgh!

[Noah Miller]

Hi -

I just deployed a system using IMAP Voicemail.  During my testing,
voicemail worked fine.  I could check vm from the phone, and the
messages would get marked as read, or I could read the messages in a
mail client, and the phone's mwi light would turn off.   Very neat.

I'm not exactly sure when things got munged up, but something broke.
I can record messages with Voicemail(), but now when I access an IMAP
mailbox using VoicemailMain(), it always says there are no messages,
even when there clearly are (unread) messages in the IMAP mailbox.

I've also got the asterisk GUI running on this system, and its status
page (retrieved via manager, I believe, or maybe voicemail show
users) shows the correct message counts.  I tried debugging manager
messages to see how it was getting the message counts, but I didn't
get any useful output.  Does anyone know a better way (any way) to
debug issues with IMAP Voicemail?  I do see an error on the CLI:

ERROR[20010]: app_voicemail.c:2026 mm_log: IMAP Error: Quota not
available on this IMAP server

Here's some background info:

Asterisk: 1.6.0.8
IMAP Server: dovecot 1.0.7
c-client: UW imap2007e

Config Files:

voicemail.conf
[general]
format = wav49
serveremail = aster...@rosecompanies.com
fromstring = ${VM_CALLERID}
emailsubject = New voicemail. Length: ${VM_DUR}
emailbody = ${VM_NAME}:\n\nYou have a new voicemail message.  You
currently have ${VM_MSGNUM} messages in your
Inbox.\n\nFrom:\t\t${VM_CALLERID}\$
maxsecs = 600
minsecs = 4
skipms = 3000
maxsilence = 10
silencethreshold = 128
maxlogins = 20
userscontext = default
imapserver = localhost
imapfolder = INBOX
authuser = asterisk
authpassword = xxx
maxgreet = 360
operator = yes
maxmessage = 300
minmessage = 4
saycid = no
sayduration = no
envelope = no
review = yes


users.conf (a typical user):
[02]
username = 02
transfer = yes
mailbox = 02
call-limit = 100
fullname = Test User
cid_number = 02
context = DLPN_MainUsers
hasvoicemail = yes
vmsecret = xxx
email =
imapuser = allison
hassip = yes
hasiax = no
host = dynamic
nat = no
hasdirectory = yes
dtmfmode = rfc2833
threewaycalling = no
callwaiting = no
hasmanager = no
hasagent = no
canreinvite = no
insecure = no
pickupgroup =
autoprov = yes
label = 02
macaddress = 0004f200
linenumber =
LINEKEYS = 1
secret = xxx
disallow = all


extensions.conf:
exten = 000,1,VoicemailMain(s${CALLERID(num)}...@default)


Thanks!
Noah

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Re: [asterisk-users] 25-50-100fxs

[Luis Morales]

Mr. ContactTel,

if you need hadware only take a look on:

http://www.telephonydepot.com/Catalog/Digium-TDM2400P/Digium-TDM2400P-Blank-Board
http://www.telephonydepot.com/Catalog/Digium-Accessories/Digium-S400M-Quad-FXS-Module
http://www.telephonydepot.com/Catalog/Digium-Accessories/Digium-1U-Patch-Panel

take a look over
http://www.audiocodes.com/product-family/cpe-gateways

Now you ever need an appliance o pc + asterisk solution.


Regards,



On Mon, Apr 6, 2009 at 1:56 PM, ContactTel Business
li...@contacttel.com wrote:
 Actually i might rephrase, i need hardware solution not pc based, no hard
 drives, no fans, no application you need to monitor, hence hardware,

 You can ignore the rest of this thread i have my info.

 Thanks


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ContactTel
 Business
 Sent: April-06-09 2:19 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] 25-50-100fxs

 Aint this based on asterisk ?

 I don't think i would use that, thanks anyway.
 And yes i know this is an asterisk list.



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales
 Sent: April-06-09 2:02 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] 25-50-100fxs

 This may be your solution.

 Regards,

 Luis Morales

 On Mon, Apr 6, 2009 at 12:23 PM, ContactTel Business
 li...@contacttel.com wrote:
 Why would i want to do that ?



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales
 Sent: April-06-09 10:27 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] 25-50-100fxs

 Take a look on xorcom solutions

 http://www.youtube.com/watch?v=qt4aPdGIvIQfeature=player_embedded

 Regards,

 Luis Morales

 On Mon, Apr 6, 2009 at 8:09 AM, ContactTel Business
 li...@contacttel.com wrote:
 Any hardware that can do 25-50-100 fxs ports trunked to sip ?



 Example one end a cat5 other end 50 RJ11’s jacks..









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 --

 
 -
 Luis Morales
 Consultor de Tecnologia
 Cel: +(58)416-4242091

 
 -
 Empieza por hacer lo necesario, luego lo que es posible... y de
 pronto estarás haciendo lo imposible

 Leonardo Da'Vinci

 
 -

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 --
 
 -
 Luis Morales
 Consultor de Tecnologia
 Cel: +(58)416-4242091
 
 -
 Empieza por hacer lo necesario, luego lo que es posible... y de
 pronto estarás haciendo lo imposible

 Leonardo Da'Vinci
 
 -

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-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible

Leonardo Da'Vinci

[asterisk-users] Sangoma and BT single lines

[Ed W]

Hi, got a Sangoma A200 with a bunch of extension cards and having real 
problems getting it to deal with a normal single BT line

Symptoms are that incoming calls are fine.  Outgoing calls ring the far 
end, BUT asterisk never sees that the call is answered (ie no message in 
the logs files saying so), as a result the remove end can hear the PBX 
side talking, but there is no audio back from the remote side to us.  
When we hangup the log files show messages thave suggest it thinks the 
line is still ringing

Comparing with another line which works fine (this is a BT multi-line 
system with what they call PBX signalling on it) I see that as soon as 
the remote end answers then asterisk gets a log message stating the same 
and audio is fine on this line


Have now spent nearly 4 months trying to get the signalling sorted on 
this line.  Most recently we requested dual signalling on the line - 
the end result is now that outbound calls work and asterisk reports that 
the phone answers, however, when you hangup the call then asterisk 
obviously gets a bunch of extra line reversals and things there is an 
immediate incoming call on the back of that outgoing call...

Please - any suggestions on how to configure a Sangoma card for use with 
a normal BT single line?

Thanks

Ed W

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[asterisk-users] Asterisk 1.6.1.0-rc4 Now Available

[Asterisk Development Team]

The Asterisk Development Team is pleased to announce the fourth release
candidate of Asterisk 1.6.1.0. Asterisk 1.6.1.0-rc4 is available for
immediate download at http://downloads.digium.com/pub/asterisk/

This release candidate improves the performance of the ast_event cache
functionality, fixes issues with rwlock corruption that would cause deadlock
like symptoms, fix queue weight behavior so that calls in low-weight queues are
not inappropriately blocked, resolves an issue with dropped calls due to SIP
re-INVITE 'glare', fixes the ability to retrieve voicemail messages from IMAP,
and several other minor issues.

This release also includes the change in the AST-2009-003 security advisory,
which can be read at http://downloads.digium.com/pub/security/AST-2009-003.html

For a full list of changes in this release candidate, please see the ChangeLog:

http://svn.digium.com/svn/asterisk/tags/1.6.1.0-rc4/ChangeLog

Issues found in this release candidate can be reported at http://bugs.digium.com

Thank you for your continued support of Asterisk!

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[asterisk-users] Asterisk 1.6.0.9 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 1.6.0.9.
Asterisk 1.6.0.9 is available for immediate download at
http://downloads.digium.com/pub/asterisk/

This release resolves a merge issue from trunk to 1.6.0.7 that caused memory
to be freed that should not be. In trunk, pkt-data is an ast_str, but in
1.6.0, it is allocated in the same chunk of memory as the sip_pkt. Only the
1.6.0 branch of Asterisk is effected by this issue.

For a summary of the changes in this release, please see the release summary:

http://svn.digium.com/svn/asterisk/tags/1.6.0.9/asterisk-1.6.0.9-summary.txt

For a full list of changes in this release, please see the ChangeLog:

http://svn.digium.com/svn/asterisk/tags/1.6.0.9/ChangeLog

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] IMAP Voicemail - can't get messages. Arrgh!

[Leif Madsen]

Noah Miller wrote:
 I just deployed a system using IMAP Voicemail.  During my testing,
 voicemail worked fine.  I could check vm from the phone, and the
 messages would get marked as read, or I could read the messages in a
 mail client, and the phone's mwi light would turn off.   Very neat.
 
 I'm not exactly sure when things got munged up, but something broke.
 I can record messages with Voicemail(), but now when I access an IMAP
 mailbox using VoicemailMain(), it always says there are no messages,
 even when there clearly are (unread) messages in the IMAP mailbox.

This appears to be the same issue as was resolved in bug 14685. If you use the 
latest version of Asterisk 1.6.0 branch then you shouldn't have that issue 
anymore.

svn co http://svn.digium.com/svn/asterisk/branches/1.6.0 asterisk-1.6.0-vanilla
cd asterisk-1.6.0-vanilla
./configure  make install

Or wait for Asterisk 1.6.0.10 which will incorporate the changes since 1.6.0.8; 
the recent 1.6.0.9 release that went out only incorporates the changes to 
1.6.0.7 plus the security release in 1.6.0.8, and a merge issue that crept into 
the 1.6.0 branch from trunk in 1.6.0.9.

Thanks!

-- 
Leif Madsen.
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk

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[asterisk-users] Hacked

[Jeremy Mann]

Just FYI:

IP address 89.248.168.176 has been trying to use the recently release SIP 
vulnerability in Asterisk to make outbound calls via our box.  They are running 
a bank account callback scam.

Jeremy Mann
Director of IT
Texas Health Management Group
Direct Line: 817-310-4956
Main Line: 817-310-4999
Helpdesk: 817-310-4999 x3
Fax: 817-310-4990
Email: jm...@txhmg.com



This e-mail, facsimile, or letter and any files or attachments transmitted with 
it contains information that is confidential and privileged. This information 
is intended only for the use of the individual(s) and entity(ies) to whom it is 
addressed. If you are the intended recipient, further disclosures are 
prohibited without proper authorization. If you are not the intended recipient, 
any disclosure, copying, printing, or use of this information is strictly 
prohibited and possibly a violation of federal or state law and regulations. If 
you have received this information in error, please notify Texas Health 
Management Group immediately at 1-817-310-4999. Texas Health Management Group, 
its subsidiaries, and affiliates hereby claim all applicable privileges related 
to this information.
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Re: [asterisk-users] ISDN Timer T309

[Martin]

Hi,

You're right. I wasn't aware of this patch getting into the code.
In the version you're running the code is already present.

The only problem I see is that some other timer kicks in here and the
T309 cannot be scheduled.

q931.c has this ...
/* For a call in Active state, activate T309 only if there is no timer
already running. */

You'd have to probably dig deeper in it to find out more. But this is
the latest explanation I see.
That would explain why the call is disconnected/hanged up right when
the alarm happens.

One way to fix it for you would be to remove the already running timer
so the T309 could be scheduled since
anyways all other timers do not matter since without T309 the call is
hanged up anyways.

Martin

On Mon, Apr 6, 2009 at 12:23 PM, Afonso Zimmermann afo...@disc-os.org wrote:
 Martin escreveu:

 Based on the Asterisk logs you posted the Asterisk doesn't have it
 implemented per:

 The implementation of timer T309 in the user side is optional

 Martin

 On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann afo...@disc-os.org
 wrote:


 Martin escreveu:

 What is the specification for T309 ? I'm too lazy to look it up.

 The default behaviour when the alarm of layer 1 (electrical T1/E1) is
 detected is to assume
 all calls dropped on both sides and that's what Asterisk does.

 The timer is simply deactivated since all the calls are supposed to
 drop. I believe that agrees with Q921/Q931 specs.

 Martin

 On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann afo...@disc-os.org
 wrote:


 Hi everione,

 I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri
 1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the
 timer fail with a telco link in this scenario:

 Telco Phone -- Telco --- Asterisk  Sip
 Phone

 When i make a call from Telco Phone to Sip Phone, the call complete, but
 when i disconnect the link and reconnect in few seconds, the Asterisk clear
 call:

 [Apr  3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 1: Red Alarm
 [Apr  3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got
 event: Alarm (4) on Primary D-channel of span 1
   == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on
 'DAHDI/1-1'
 [Apr  3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No
 D-channels available!  Using Primary channel 16 as D-channel anyway!
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 2: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 2: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 3: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 3: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 4: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 4: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 5: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 5: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 6: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 6: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 7: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 7: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 8: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 8: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 9: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 9: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 10: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 10: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 11: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 11: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on 

Re: [asterisk-users] Hacked

[ContactTel Business]

http://www.websiteoutlook.com/www.songania.com

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy Mann
Sent: April-06-09 3:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Hacked

 

Just FYI:

 

IP address 89.248.168.176 has been trying to use the recently release SIP
vulnerability in Asterisk to make outbound calls via our box.  They are
running a bank account callback scam.

 

Jeremy Mann

Director of IT

Texas Health Management Group

Direct Line: 817-310-4956

Main Line: 817-310-4999

Helpdesk: 817-310-4999 x3

Fax: 817-310-4990

Email: jm...@txhmg.com

 

 

  _  

This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and
entity(ies) to whom it is addressed. If you are the intended recipient,
further disclosures are prohibited without proper authorization. If you are
not the intended recipient, any disclosure, copying, printing, or use of
this information is strictly prohibited and possibly a violation of federal
or state law and regulations. If you have received this information in
error, please notify Texas Health Management Group immediately at
1-817-310-4999. Texas Health Management Group, its subsidiaries, and
affiliates hereby claim all applicable privileges related to this
information.

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Re: [asterisk-users] Hacked

[Jeff LaCoursiere]

Ok, I'll bite.  What does websiteoutlook have to do with it?

The IP mentioned is in the Netherlands:

% Information related to '89.248.168.0 - 89.248.168.255'

inetnum:89.248.168.0 - 89.248.168.255
netname:NL-ECATEL
descr:  AS29073, Ecatel LTD
country:NL
admin-c:EL25-RIPE
tech-c: EL25-RIPE
status: ASSIGNED PA
mnt-by: ECATEL-MNT
mnt-lower:  ECATEL-MNT
mnt-routes: ECATEL-MNT
source: RIPE # Filtered

role:   Ecatel LTD
address:Gyroscoopweg 2F
address:1042AB Amsterdam
address:Netherlands
abuse-mailbox:  ab...@ecatel.net
admin-c:EL25-RIPE
tech-c: EL25-RIPE
nic-hdl:EL25-RIPE
source: RIPE # Filtered

% Information related to '89.248.168.0/24as29073'

route:  89.248.168.0/24
descr:  AS29073 route object
origin: as29073
mnt-by: ECATEL-MNT
source: RIPE # Filtered


j

On Mon, 6 Apr 2009, ContactTel Business wrote:

 http://www.websiteoutlook.com/www.songania.com







 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy Mann
 Sent: April-06-09 3:55 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [asterisk-users] Hacked



 Just FYI:



 IP address 89.248.168.176 has been trying to use the recently release SIP
 vulnerability in Asterisk to make outbound calls via our box.  They are
 running a bank account callback scam.



 Jeremy Mann

 Director of IT

 Texas Health Management Group

 Direct Line: 817-310-4956

 Main Line: 817-310-4999

 Helpdesk: 817-310-4999 x3

 Fax: 817-310-4990

 Email: jm...@txhmg.com





  _

 This e-mail, facsimile, or letter and any files or attachments transmitted
 with it contains information that is confidential and privileged. This
 information is intended only for the use of the individual(s) and
 entity(ies) to whom it is addressed. If you are the intended recipient,
 further disclosures are prohibited without proper authorization. If you are
 not the intended recipient, any disclosure, copying, printing, or use of
 this information is strictly prohibited and possibly a violation of federal
 or state law and regulations. If you have received this information in
 error, please notify Texas Health Management Group immediately at
 1-817-310-4999. Texas Health Management Group, its subsidiaries, and
 affiliates hereby claim all applicable privileges related to this
 information.



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Re: [asterisk-users] Hacked

[ContactTel Business]

ping www.songania.com
PING www.songania.com (89.248.168.176) 56(84) bytes of data.
64 bytes from 89.248.168.176: icmp_seq=1 ttl=49 time=131 ms

If you clicked on it you would of seen it shows info on the domain, that is
hosted on it.. ill bite back ;)

Then on bottom.. Owned By Al-Sharif

Al-sharif ? rings a bell.. but who knows.. iptables --block all the worls
minus what you want.. 




-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: April-06-09 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hacked


Ok, I'll bite.  What does websiteoutlook have to do with it?

The IP mentioned is in the Netherlands:

% Information related to '89.248.168.0 - 89.248.168.255'

inetnum:89.248.168.0 - 89.248.168.255
netname:NL-ECATEL
descr:  AS29073, Ecatel LTD
country:NL
admin-c:EL25-RIPE
tech-c: EL25-RIPE
status: ASSIGNED PA
mnt-by: ECATEL-MNT
mnt-lower:  ECATEL-MNT
mnt-routes: ECATEL-MNT
source: RIPE # Filtered

role:   Ecatel LTD
address:Gyroscoopweg 2F
address:1042AB Amsterdam
address:Netherlands
abuse-mailbox:  ab...@ecatel.net
admin-c:EL25-RIPE
tech-c: EL25-RIPE
nic-hdl:EL25-RIPE
source: RIPE # Filtered

% Information related to '89.248.168.0/24as29073'

route:  89.248.168.0/24
descr:  AS29073 route object
origin: as29073
mnt-by: ECATEL-MNT
source: RIPE # Filtered


j

On Mon, 6 Apr 2009, ContactTel Business wrote:

 http://www.websiteoutlook.com/www.songania.com







 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy Mann
 Sent: April-06-09 3:55 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [asterisk-users] Hacked



 Just FYI:



 IP address 89.248.168.176 has been trying to use the recently release SIP
 vulnerability in Asterisk to make outbound calls via our box.  They are
 running a bank account callback scam.



 Jeremy Mann

 Director of IT

 Texas Health Management Group

 Direct Line: 817-310-4956

 Main Line: 817-310-4999

 Helpdesk: 817-310-4999 x3

 Fax: 817-310-4990

 Email: jm...@txhmg.com





  _

 This e-mail, facsimile, or letter and any files or attachments transmitted
 with it contains information that is confidential and privileged. This
 information is intended only for the use of the individual(s) and
 entity(ies) to whom it is addressed. If you are the intended recipient,
 further disclosures are prohibited without proper authorization. If you
are
 not the intended recipient, any disclosure, copying, printing, or use of
 this information is strictly prohibited and possibly a violation of
federal
 or state law and regulations. If you have received this information in
 error, please notify Texas Health Management Group immediately at
 1-817-310-4999. Texas Health Management Group, its subsidiaries, and
 affiliates hereby claim all applicable privileges related to this
 information.



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[asterisk-users] Zaptel Config

[Torintino T]

Is there different points in the zaptel configuration according to each country?

Thanks.

_
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Re: [asterisk-users] Zaptel Config

[Danny Nicholas]

I'd read this article
(http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf) but as I see it,
you only have 2 lines in zaptel.conf for country specification; the rest of
the lifting is done in Zapata.conf.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T
Sent: Monday, April 06, 2009 3:55 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Zaptel Config

 

Is there different points in the zaptel configuration according to each
country?

Thanks.

  _  

What can you do with the new Windows Live? Find
http://www.microsoft.com/windows/windowslive/default.aspx  out

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Re: [asterisk-users] IMAP Voicemail - can't get messages. Arrgh!

[Noah Miller]

 I'm not exactly sure when things got munged up, but something broke.
 I can record messages with Voicemail(), but now when I access an IMAP
 mailbox using VoicemailMain(), it always says there are no messages,
 even when there clearly are (unread) messages in the IMAP mailbox.

 This appears to be the same issue as was resolved in bug 14685. If you use the
 latest version of Asterisk 1.6.0 branch then you shouldn't have that issue 
 anymore.

Aha!  Thanks, Leif!  I'm not insane.  OK, well, maybe I am.  I didn't
find that bug.  I think I'm going to bite the bullet and go with
1.6.1.0-rc4.  Some of those items in the 1.6.1.0rc4 changelog just
look to good to be passed up (or too scary to ignore).


Thanks,
Noah

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Re: [asterisk-users] Zaptel Config

[Torintino T]

so they are only.

loazone
and
defaultzone

thanks.
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Mon, 6 Apr 2009 16:01:25 -0500
Subject: Re: [asterisk-users] Zaptel Config






















I’d read this article 
(http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf)
but as I see it, you only have 2 lines in zaptel.conf for country
specification; the rest of the lifting is done in Zapata.conf.

 









From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T

Sent: Monday, April 06, 2009 3:55
PM

To: asterisk-users@lists.digium.com

Subject: [asterisk-users] Zaptel
Config



 

Is there different points in the
zaptel configuration according to each country?



Thanks.







What can you do with the new Windows Live? Find
out


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[asterisk-users] One way AUDIO

[David @ULC]

I have a server with 2 Lan Cards.

Now, when I am trying to make calls using Local Lan, its One way Audio which
means customer cant hear me but if I use Static IP with Wan Connection, it
works perfectly.

I changed the network from loc1 to loc2 but its same.

I tried changing Ethernet Card but no use.

What could be the Issue ?
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Re: [asterisk-users] One way AUDIO

[David @ULC]

Can it be that any Port got blocked ?

On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote:


 I have a server with 2 Lan Cards.

 Now, when I am trying to make calls using Local Lan, its One way Audio
 which means customer cant hear me but if I use Static IP with Wan
 Connection, it works perfectly.

 I changed the network from loc1 to loc2 but its same.

 I tried changing Ethernet Card but no use.

 What could be the Issue ?

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Re: [asterisk-users] One way AUDIO

[Giancarlo Rubio]

How  tcpdump on interface show??

2009/4/6 David @ULC ucoms2...@gmail.com:

 Can it be that any Port got blocked ?

 On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote:

 I have a server with 2 Lan Cards.

 Now, when I am trying to make calls using Local Lan, its One way Audio
 which means customer cant hear me but if I use Static IP with Wan
 Connection, it works perfectly.

 I changed the network from loc1 to loc2 but its same.

 I tried changing Ethernet Card but no use.

 What could be the Issue ?


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-- 
Giancarlo Rubio

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Re: [asterisk-users] One way AUDIO

[David @ULC]

Few Running figures !!

On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote:


 I have a server with 2 Lan Cards.

 Now, when I am trying to make calls using Local Lan, its One way Audio
 which means customer cant hear me but if I use Static IP with Wan
 Connection, it works perfectly.

 I changed the network from loc1 to loc2 but its same.

 I tried changing Ethernet Card but no use.

 What could be the Issue ?

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Re: [asterisk-users] Hacked

[Martin]

Can you give more information about this vulnerability ?

Martin

On Mon, Apr 6, 2009 at 2:55 PM, Jeremy Mann jm...@txhmg.com wrote:
 Just FYI:



 IP address 89.248.168.176 has been trying to use the recently release SIP
 vulnerability in Asterisk to make outbound calls via our box.  They are
 running a bank account callback scam.



 Jeremy Mann

 Director of IT

 Texas Health Management Group

 Direct Line: 817-310-4956

 Main Line: 817-310-4999

 Helpdesk: 817-310-4999 x3

 Fax: 817-310-4990

 Email: jm...@txhmg.com



 
 This e-mail, facsimile, or letter and any files or attachments transmitted
 with it contains information that is confidential and privileged. This
 information is intended only for the use of the individual(s) and
 entity(ies) to whom it is addressed. If you are the intended recipient,
 further disclosures are prohibited without proper authorization. If you are
 not the intended recipient, any disclosure, copying, printing, or use of
 this information is strictly prohibited and possibly a violation of federal
 or state law and regulations. If you have received this information in
 error, please notify Texas Health Management Group immediately at
 1-817-310-4999. Texas Health Management Group, its subsidiaries, and
 affiliates hereby claim all applicable privileges related to this
 information.

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[asterisk-users] Australian NBN network announced

[Dean Collins]

First posted at:
http://deancollinsblog.blogspot.com/2009/04/australian-nbn-network.html

 

 

 

 

 

 

You ripper,
http://www.crn.com.au/News/100466,government-announces-nbn-plans.aspx

This is exactly how fiber should be - a shared wholesale resource just
like water to ensure retail channels can deliver value add in content
and services;

- Not 'outspending on infrastructure' disadvantaging the carrier
or
- Securing 'private monopolies' that disadvantage the customer.


Lets hope this smashes the existing theory about how networks should be
built.




Regards,

Dean Collins
Cognation Inc
d...@cognation.net
mailto:d...@cognation.net +1-212-203-4357   New York
+61-2-9016-5642   (Sydney in-dial).
+44-20-3129-6001 (London in-dial).





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Re: [asterisk-users] fail to retrieve the calling party information

[Rilawich Ango]

Thanks.  Let me try it.

On Tue, Apr 7, 2009 at 12:23 AM, Martin asteriskl...@callthem.info wrote:
 That's because you have to create a user account in sip.conf ... +
 Asterisk is sensitive about it.
 What should help is if you register the phone with that sip account first.

 Martin

 On Mon, Apr 6, 2009 at 5:00 AM, Rilawich Ango maillist...@gmail.com wrote:
 HI,

  Recently, I found that asterisk fail to get the correct context of
 the sip phone.  Below is the configuration and the log message.  In
 the log message, asterisk fail to identify the calling party.  As a
 result, it use a default context instead of int.  Anyone know why and
 how to fix it?

 (testing environment)
 asterisk 1.4.22  1.4.24
 asterisk-addon-1.4.7

 Setting
 name=123
 context=int

 [Apr  6 17:53:12] NOTICE[4913] chan_sip.c: Call from '' to extension
 '5544' rejected because extension not found.

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Re: [asterisk-users] fail to retrieve the calling party information

[ContactTel Business]

Yeah some devices use callerid as user which is xxx in  x...@deviceip
So if you see : chan_sip.c: Call from '1231231234' to extension
 '5544' rejected because extension not found.
Then adding in the [user] stanza

user=foobar
fromuser=foobar
insecure=very ( or port,invite if that still alive)


Will make sure it can auth ok.. 



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rilawich Ango
Sent: April-06-09 10:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] fail to retrieve the calling party information

Thanks.  Let me try it.

On Tue, Apr 7, 2009 at 12:23 AM, Martin asteriskl...@callthem.info wrote:
 That's because you have to create a user account in sip.conf ... +
 Asterisk is sensitive about it.
 What should help is if you register the phone with that sip account first.

 Martin

 On Mon, Apr 6, 2009 at 5:00 AM, Rilawich Ango maillist...@gmail.com
wrote:
 HI,

  Recently, I found that asterisk fail to get the correct context of
 the sip phone.  Below is the configuration and the log message.  In
 the log message, asterisk fail to identify the calling party.  As a
 result, it use a default context instead of int.  Anyone know why and
 how to fix it?

 (testing environment)
 asterisk 1.4.22  1.4.24
 asterisk-addon-1.4.7

 Setting
 name=123
 context=int

 [Apr  6 17:53:12] NOTICE[4913] chan_sip.c: Call from '' to extension
 '5544' rejected because extension not found.

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Re: [asterisk-users] Australian NBN network announced

[Steve Edwards]

On Mon, 6 Apr 2009, Dean Collins wrote:

 http://www.crn.com.au/News/100466,government-announces-nbn-plans.aspx

 This is exactly how fiber should be - a shared wholesale resource just 
 like water to ensure retail channels can deliver value add in content 
 and services;

 - Not 'outspending on infrastructure' disadvantaging the carrier or - 
 Securing 'private monopolies' that disadvantage the customer.

 Lets hope this smashes the existing theory about how networks should be 
 built.

They must have a different kind of government down under ;)

Thanks in advance,

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

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[asterisk-users] asterisk and patton

[mahboob zaman]

Hellow

Can any body helps how can interfacing between asterisk and patton media
getway.

Thanks
mahboob
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Re: [asterisk-users] asterisk and patton

[Alex Balashov]

Interface in what manner?

mahboob zaman wrote:

 
 Hellow
  
 Can any body helps how can interfacing between asterisk and patton media 
 getway.
  
 Thanks
 mahboob
  
 
 
 
 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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