Re: [asterisk-users] T.38 on a MAX/Lucent/Ascend TNT

2010-06-25 Thread Miguel Amez
Hi!

I have probed t38modem several times without success.
The most I get was sending with t38modem and terminate faxes on a third-part
reseller, but with low bandwith sync between t38modem and my asterisk. it
syncs at 2400 bpps.
So If we (you and me) could get a working configuration for this, It will be
great.
I've wrote this list several times asking for this issue without any answer.

If you wonna collaborate, please tell me.

Regards,
Miguel Amez

2010/6/24 Ben Winslow winsl...@pa.net

 Hello folks,

 I've been trying to get T.38 over SIP working with calls terminated by a
 MAX/Lucent/Ascent TNT.  As far as I can tell, SIP and T.38 are actually
 working perfectly; however, I can't get the TNT to properly terminate a
 FAX call.  Does anyone have a working configuration for SIP and T.38 for
 calls from a TNT or APX?

 Here's a brief description/diagram of my test setup:

 Laptop --RS232-- Modem --POTS-- Channel bank (ADIT 600) --CAS T1--
 TNT --Ethernet/SIP-- Asterisk --SIP-- t38modem.

 The TNT is running TAOS 11.0.4 with both the SIP and realtime fax
 features licensed.  I've tried using several modems, and with each I can
 establish a data call without any problems (although at a maximum of
 31200bps/3429 baud using a 33.6 modem, for reasons I haven't dug into
 yet.)  Whenever I try to send a fax from the laptop, however, the call
 always seems to fail in the first HDLC phase (phase B) with either a
 timeout or error 23 (COMREC invalid command received.)  The modem is
 connected directly to the channel bank and the channel bank is connected
 directly to the TNT in an attempt to reduce the number of variables.

 With my current configuration, the call to Asterisk will come up as a
 voice call, then be dumped into t38modem when Asterisk receives the
 1100Hz CNG tone from the sending modem.  At that point, the call is
 dumped into t38modem which re-invites the TNT with the T.38 options, and
 the TNT usually sends a T.38 t30-ind of 0x00 (no-signal) or 0x3a (???),
 although I do occasionally see more promising messages like
 v29-7200-training or t30-data/hdlc-fcs-ok.  Shortly thereafter, the
 dialing modem will give up and terminate the call.

 If I try dumping the same call into iaxmodem instead of t38modem, the
 call actually progresses further -- the real modem receives and decodes
 the HDLC CSI/DIS from iaxmodem, but the high-speed trainup always fails
 for calls coming from the TNT.

 Does anyone have any advice or suggestions?  Has anyone actually made
 T.38 work with one of these devices running ANY TAOS version?

 Thanks,
 --
 Ben Winslow winsl...@pa.net

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[asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?

2010-06-25 Thread Gilles
Hello

About every three months, my dad's little Asterisk server that handles
his business phone line with an OpenVox PCI card stops taking calls.

To check if it's the cause, I'd like to run a CRON job every night to
restart Zaptel and Asterisk.

Before I go ahead, I'd like to know if I can just send the following
commands, or if there are issues I should know about:

/usr/local/etc/rc.d/asterisk stop
/usr/local/etc/rc.d/zaptel stop
/usr/local/etc/rc.d/zaptel start
/usr/local/etc/rc.d/asterisk start

Or, as I suspect, if there's already a good watchguard applet
available, I could use that instead.

FWIW, it's a FreeBSD 6.3 host, running Zaptel 1.4.0-BSD and Asterisk
1.4.21.2.

Thank you.


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[asterisk-users] G729 license key registration

2010-06-25 Thread Kiss András
Hi,

I have trouble re-registering a G729 license for Asterisk (bought 6 years ago)
My license looks like: 10D2X----X

Tried to re-register the codec according to the
http://downloads.digium.com/pub/telephony/codec_g729/README document,
but the register failed with this error message:

You selected 5, G.729 Codec
Please enter your Key-ID: 10D2X----X
This product key cannot be registered!  Please verify you entered the
correct product key.
Server response: ERR - Invalid prefix, should be 'G729'

The program can communicate correctly with digium`s server.
Tried with the G729 prefix, but it seems completely wrong:

You selected 5, G.729 Codec
Please enter your Key-ID: G729-10D2X----X
This product key cannot be registered!  Please verify you entered the
correct product key.
Server response: 404 - Key not found.


Any suggestions?

- Andras

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Re: [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?

2010-06-25 Thread Randy R
On Fri, Jun 25, 2010 at 8:59 AM, Gilles codecompl...@free.fr wrote:
 Hello

 About every three months, my dad's little Asterisk server that handles
 his business phone line with an OpenVox PCI card stops taking calls.

 To check if it's the cause, I'd like to run a CRON job every night to
 restart Zaptel and Asterisk.

IMO, if it's a business phone, you'd do well to just reboot it at 3AM
once a week or once a month or some interval that you're comfortable
with. We used to do this for a similar reason.

/r

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Re: [asterisk-users] G729 license key registration

2010-06-25 Thread Remco Bressers
On 06/25/2010 09:48 AM, Kiss András wrote:
 You selected 5, G.729 Codec
 Please enter your Key-ID: G729-10D2X----X
 This product key cannot be registered!  Please verify you entered the
 correct product key.
 Server response: 404 - Key not found.
 
 Any suggestions?

How about contacting Digium about this?

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Re: [asterisk-users] SPA8000 outbound CID problem

2010-06-25 Thread dotnetdub
On 24 June 2010 19:54, Mark G. Thomas m...@misty.com wrote:

 Hi,

 I'm trying to configure a Linksys/Cisco SPA8000 talking SIP to
 both a local Asterisk server and also with a trunk directly to
 a VOIP provider. Everything works great, except I'm having a problem
 setting the outbound caller ID to a value different from the
 SIP username/authname.

 The SPA8000 has SIP setting for Display Name, User ID, Password,
 and Auth ID, as well as a Use Auth ID checkbox. It's running 6.1.3
 firmware, which looks to be the latest, and supports SIP trunking, though
 even if I don't use trunking, I have the same obstacle if I configure it
 per-line instead of per-trunk.

 Inbound CID works fine. When VOIP calls come in via the provider or
 Asterisk, the SPA generates CID on it's analog ports.

 The problem is that the outbound caller ID number seems to come from
 the SIP User ID setting, which is also the SIP authentication name.
 If I instead put the SIP account id into the Auth ID field and check
 the Use Auth ID box, Asterisk reports:

  Registration from 'John Smith 
 sip:jsm...@our.sip.gateway.comsip%3ajsm...@our.sip.gateway.com'
 failed for
  '1.2.3.4' - Username/auth name mismatch.

 Sure, I can overide the CID number on our Asterisk server, but I don't
 have that ability with the VOIP provider's Asterisk server. The outbound
 caller ID always looks like John Smith jsmith instead of
 John Smith 211212 no matter how I try to set these fields.

 I take it the SIP username and auth name need to match, so that leaves me
 with the question of how to configure a CID number that doesn't necessarily
 match the SIP user/auth name. Is this a limitation of this device, or
 is there some other option I'm overlooking?

 Mark


 Ask your upstream provider if they support remote party ID. IF they do you
can set sendrpid=yes in your sip.conf and set your outbound CID on an
extension or trunk level.

HTH
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Re: [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?

2010-06-25 Thread Tzafrir Cohen
On Fri, Jun 25, 2010 at 08:59:32AM +0200, Gilles wrote:
 Hello
 
 About every three months, my dad's little Asterisk server that handles
 his business phone line with an OpenVox PCI card stops taking calls.
 
 To check if it's the cause, I'd like to run a CRON job every night to
 restart Zaptel and Asterisk.

That does not really check if that is the problem. Mind giving more
information as for the nature of the problem?

 
 Before I go ahead, I'd like to know if I can just send the following
 commands, or if there are issues I should know about:
 
 /usr/local/etc/rc.d/asterisk stop
 /usr/local/etc/rc.d/zaptel stop
 /usr/local/etc/rc.d/zaptel start
 /usr/local/etc/rc.d/asterisk start
 
 Or, as I suspect, if there's already a good watchguard applet
 available, I could use that instead.
 
 FWIW, it's a FreeBSD 6.3 host, running Zaptel 1.4.0-BSD and Asterisk
 1.4.21.2.

'dahdi restart' has become more relaible as of 1.4.22 ...

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Re: [asterisk-users] OT: Bandwidth calculations

2010-06-25 Thread Chris Bagnall
 ISP 10% rule is what you are asking about
 expected that average usage is 10% of total subscribers with bursts 
 higher

But remember to plan well for those bursts and ensure you have sufficient 
excess capacity. Certain events can have a significant effect on your burst 
pattern: some fellows are kicking a ball around in South Africa for three 
weeks, which is having an understandable effect on bandwidth usage globally. 
Same happened a couple of years ago during the Olympics.

Regards,

Chris

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Re: [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?

2010-06-25 Thread Gilles
On Fri, 25 Jun 2010 09:53:34 +0200, Randy R randulo2...@gmail.com
wrote:
IMO, if it's a business phone, you'd do well to just reboot it at 3AM
once a week or once a month or some interval that you're comfortable
with. We used to do this for a similar reason.

Right, but he won't remember to do this, and I don't want to do it
myself :-)

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Re: [asterisk-users] OT: Bandwidth calculations

2010-06-25 Thread Gareth Blades
Randy R wrote:
 Hi,
 
 I know some of you are very experienced  as to the working of
 networks. I wondered whether there is some accepted way of determining
 bandwidth needs based on the network traffic over time. For example,
 looking at the figures for the network traffic through the server
 interface, we have hourly, daily and monthly figures. If everything
 were linear, taking the hourly figure and dividing it by 3600 (or the
 daily by divided 3600*24) would give us the required bps, but this
 average is pretty meaningless.
 
 Those of you who have experience and education in this area, where can
 I look for guidance (links?) and do you have any rules of thumb you'd
 care to share? I'm actually looking for this for a web server not
 VoiP, but any info is welcome.
 
 It seems obvious to me that taking the per second average and
 multiplying it by some kind of seat of the pants number must give a
 decent idea? WHat is that magic number?
 
 Thanks in advance for any ideas.
 
 /r
 

It depends on the charging structure your ISP is using.

 From an ISP point of view when they purchase transit links all pricing 
is done on the 95th percentile basis. So for example you might purchase 
a 50Mbps connection over a 100Mbps interface. You would get a guaranteed 
50Mbps but could burst up to the full interface speed.
Bandwidth measurements are taken normally every 5 or 10 minutes. The 
95th percentile is calculated by discarding the top 5% results and the 
next highest reading is the 95th percentile. If this is below the 50Mbps 
then all is fine otherwise you will receive a bill for overusage.

For a web server this is probably a good start but would depend on how 
spiky your bandwidth graphs are. You might want to lower the speed if 
you tend to have lots of people downloading on a particular day of the 
week for example as the 95th percentile would be much greater than your 
average bandwidth.

If you just look at your bandwidth graphs you will probably get the best 
idea of a suitable figure.

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Re: [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?

2010-06-25 Thread Gilles
On Fri, 25 Jun 2010 11:43:04 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
That does not really check if that is the problem. Mind giving more
information as for the nature of the problem?

I don't have more information. Could just be the Zaptel driver working
with the OpenVoice card, in which case unloading/reloading Zaptel
could solve the problem.

'dahdi restart' has become more relaible as of 1.4.22 ...

If possible, I'd rather not uninstall Zaptel and install Dahdi because
the system works OK the rest of the time (installed about 3 years
ago).

So I guess I'll just send the stop/start commands and see how it goes.

Thank you.

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Re: [asterisk-users] OT: Bandwidth calculations

2010-06-25 Thread Randy R
On Fri, Jun 25, 2010 at 11:36 AM, Gareth Blades
list-aster...@skycomuk.com wrote:
 For a web server this is probably a good start but would depend on how
 spiky your bandwidth graphs are. You might want to lower the speed if

The max in the past 24 was 140MB an hour, but I've seen up to 240MB in
an hour. This happens maybe once in a day. The average is more around
60-75 MB in an hour. The interface is currently at 30Mbps. Obviously,
if 10 people look at the same time at some fat page, the server will
be brought to its knees. The video is all on a separate CDN and I'm
trying to get the people to add the larger images to that.

/r

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Re: [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?

2010-06-25 Thread Randy R
On Fri, Jun 25, 2010 at 11:44 AM, Gilles codecompl...@free.fr wrote:
 On Fri, 25 Jun 2010 09:53:34 +0200, Randy R randulo2...@gmail.com
once a week or once a month or some interval that you're comfortable
with. We used to do this for a similar reason.

 Right, but he won't remember to do this, and I don't want to do it
 myself :-)

No I meant as a CRON job! No one will be calling at 3AM, there's ample
time to reboot once a month for example.

Of course as Tzafrir points out, this doesn't tell you what's wrong.
What it does do is lessen any possible problems with memory leaks,
etc.

/r

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Re: [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?

2010-06-25 Thread Gilles
On Fri, 25 Jun 2010 12:09:09 +0200, Randy R randulo2...@gmail.com
wrote:
No I meant as a CRON job! No one will be calling at 3AM, there's ample
time to reboot once a month for example.

Sorry for the misunderstanding. So I can just run reboot from a CRON
job then.

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Re: [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?

2010-06-25 Thread Hugo Serrano
On 25/06/10 10:44, Gilles wrote:
 On Fri, 25 Jun 2010 09:53:34 +0200, Randy Rrandulo2...@gmail.com
 wrote:

 IMO, if it's a business phone, you'd do well to just reboot it at 3AM
 once a week or once a month or some interval that you're comfortable
 with. We used to do this for a similar reason.
  
 Right, but he won't remember to do this, and I don't want to do it
 myself :-)


You can set it up.

#crontab -e

0 3 15 * * /sbin/reboot


This examples should reboot the server every day 15 of any month at 3 a.m.

0 - minutes
3 - hours
15 - Day of the Month
* - Month
* - Day of the week
/sbin/reboot - Comand

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Javali
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Re: [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?

2010-06-25 Thread Gareth Blades
Hugo Serrano wrote:
 On 25/06/10 10:44, Gilles wrote:
 On Fri, 25 Jun 2010 09:53:34 +0200, Randy Rrandulo2...@gmail.com
 wrote:

 IMO, if it's a business phone, you'd do well to just reboot it at 3AM
 once a week or once a month or some interval that you're comfortable
 with. We used to do this for a similar reason.
  
 Right, but he won't remember to do this, and I don't want to do it
 myself :-)


 You can set it up.
 
 #crontab -e
 
 0 3 15 * * /sbin/reboot
 
 
 This examples should reboot the server every day 15 of any month at 3 a.m.
 
 0 - minutes
 3 - hours
 15 - Day of the Month
 * - Month
 * - Day of the week
 /sbin/reboot - Comand
 

If you are going to reboot the server regularly then make sure and 
system updates are set to not automatically install new kernel versions.
Otherwise if you get a kernel update and reboot zaptel/dahdi wont load 
until you recompile it.

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Re: [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?

2010-06-25 Thread Randy R
On Fri, Jun 25, 2010 at 12:10 PM, Gilles codecompl...@free.fr wrote:
 Sorry for the misunderstanding. So I can just run reboot from a CRON
 job then.

From root's cron, yes

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Re: [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?

2010-06-25 Thread Tzafrir Cohen
On Fri, Jun 25, 2010 at 11:46:37AM +0200, Gilles wrote:

 'dahdi restart' has become more relaible as of 1.4.22 ...
 
 If possible, I'd rather not uninstall Zaptel and install Dahdi because
 the system works OK the rest of the time (installed about 3 years
 ago).
 
 So I guess I'll just send the stop/start commands and see how it goes.

Asterisk 1.4.x works with Zaptel as well.

But yes, this means an upgrade of Asterisk, and maybe you'd like to
avoid that.

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[asterisk-users] Is there a default dial plan that is not in extention.conf?

2010-06-25 Thread Eyal Goltzman
Hi,

 

I have a trivial peace of dialplan for exten 100. I try to change it to _1XX
and the asterisk act according to a different (Default??) dial plan and not
the one I want? Is that possible? Where is the other dialplan sits? In my
extention.conf I can't see something that look like what asterisk is
dialing.

How can I trace\debug my dialplan?

 

Thanks,

 

Eyal

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Re: [asterisk-users] Is there a default dial plan that is not in extention.conf?

2010-06-25 Thread Geraint Lee
try looking in extensions.ael

On 25 June 2010 12:25, Eyal Goltzman egoltz...@gmail.com wrote:

  Hi,



 I have a trivial peace of dialplan for exten 100. I try to change it to
 _1XX and the asterisk act according to a different (Default??) dial plan and
 not the one I want? Is that possible? Where is the other dialplan sits? In
 my extention.conf I can't see something that look like what asterisk is
 dialing.

 How can I trace\debug my dialplan?



 Thanks,



 Eyal

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Re: [asterisk-users] Is there a default dial plan that is not in extention.conf?

2010-06-25 Thread Doug Lytle
Eyal Goltzman wrote:

 Hi,

 I have a trivial peace of dialplan for exten 100. I try to change it 
 to _1XX and the asterisk act according to a different (Default??) dial 
 plan and not the one I want? Is that


Does dialplan show output more then expected?

You can have more then 1 file linked for your dialplan.  Using the 
#include statement.

Doug

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Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] Is there a default dial plan that is not in extention.conf?

2010-06-25 Thread Paul Belanger
On Fri, Jun 25, 2010 at 7:25 AM, Eyal Goltzman egoltz...@gmail.com wrote:
 How can I trace\debug my dialplan?

*CLI dialplan show 1...@context

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Re: [asterisk-users] T.38 on a MAX/Lucent/Ascend TNT

2010-06-25 Thread JR Richardson
 Date: Thu, 24 Jun 2010 15:32:39 -0400
 From: Ben Winslow winsl...@pa.net
 Subject: [asterisk-users] T.38 on a MAX/Lucent/Ascend TNT
 To: asterisk-users@lists.digium.com
 Message-ID: 4c23b2d7.9090...@pa.net
 Content-Type: text/plain; charset=ISO-8859-1; format=flowed
 
 Hello folks,
 
 I've been trying to get T.38 over SIP working with calls terminated by a
 MAX/Lucent/Ascent TNT.  As far as I can tell, SIP and T.38 are actually
 working perfectly; however, I can't get the TNT to properly terminate a
 FAX call.  Does anyone have a working configuration for SIP and T.38 for
 calls from a TNT or APX?
 
 Here's a brief description/diagram of my test setup:
 
 Laptop --RS232-- Modem --POTS-- Channel bank (ADIT 600) --CAS T1--
 TNT --Ethernet/SIP-- Asterisk --SIP-- t38modem.
 
 The TNT is running TAOS 11.0.4 with both the SIP and realtime fax
 features licensed.  I've tried using several modems, and with each I can
 establish a data call without any problems (although at a maximum of
 31200bps/3429 baud using a 33.6 modem, for reasons I haven't dug into
 yet.)  Whenever I try to send a fax from the laptop, however, the call
 always seems to fail in the first HDLC phase (phase B) with either a
 timeout or error 23 (COMREC invalid command received.)  The modem is
 connected directly to the channel bank and the channel bank is connected
 directly to the TNT in an attempt to reduce the number of variables.
 
 With my current configuration, the call to Asterisk will come up as a
 voice call, then be dumped into t38modem when Asterisk receives the
 1100Hz CNG tone from the sending modem.  At that point, the call is
 dumped into t38modem which re-invites the TNT with the T.38 options, and
 the TNT usually sends a T.38 t30-ind of 0x00 (no-signal) or 0x3a (???),
 although I do occasionally see more promising messages like
 v29-7200-training or t30-data/hdlc-fcs-ok.  Shortly thereafter, the
 dialing modem will give up and terminate the call.
 
 If I try dumping the same call into iaxmodem instead of t38modem, the
 call actually progresses further -- the real modem receives and decodes
 the HDLC CSI/DIS from iaxmodem, but the high-speed trainup always fails
 for calls coming from the TNT.
 
 Does anyone have any advice or suggestions?  Has anyone actually made
 T.38 work with one of these devices running ANY TAOS version?
 
 Thanks,
 --
 Ben Winslow winsl...@pa.net
 

What version of Asterisk are you running?  I've struggled to get this
working as well but with TNT TOAS version 14 and Asterisk 6.1.x I managed to
get reliable T38 faxes one-way: from fax to PSTN

Fax machineSIP T38 ATAAsteriskSIP T38MAX TNTPSTN PRI

The other way does not work, the call doesn't switch to T38 and I haven't
had time to investigate too deeply.

Fax handling in TOAS was greatly improved in 14.0 version, I would suggest
you upgrade to that if you can and start there.

JR


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Re: [asterisk-users] Is there a default dial plan that is not in extention.conf?

2010-06-25 Thread Tzafrir Cohen
On Fri, Jun 25, 2010 at 02:25:38PM +0300, Eyal Goltzman wrote:
 Hi,
 
  
 
 I have a trivial peace of dialplan for exten 100. I try to change it to _1XX
 and the asterisk act according to a different (Default??) dial plan and not
 the one I want? Is that possible? Where is the other dialplan sits? In my
 extention.conf I can't see something that look like what asterisk is
 dialing.
 
 How can I trace\debug my dialplan?

To see where it comes from, run in the Asterisk CLI:

  dialplan show context

or:

  dialplan show exten@context

Here is a partial output from 'dialplan show' here, that shows all of
them (but is normally overly long)

[ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ]
  's' =1. NoOp() [app_queue]

[ Context 'parkedcalls' created by 'features' ]
  '700' =  1. Park() [features]

[ Context 'app_dial_gosub_virtual_context' created by 'app_dial' ]
  's' =1. NoOp() [app_dial]

[ Context 'from-pstn' created by 'pbx_config' ]
  '_X.' =  1. Answer()   [pbx_config]
2. Playback(demo-instruct)[pbx_config]
3. Hangup()   [pbx_config]

[ Context 'ael-dundi-e164' created by 'pbx_ael' ]
  's' =1. MSet(LOCAL(exten)=${ARG1}) [pbx_ael]
2. Goto(${exten},1)   [pbx_ael]
3. Return()   [pbx_ael]


'pbx_config' is dialplan that was generated from your extensions.conf. 
'pbx_ael' is dialplan that was generated from extensions.ael.
Various other modules include their own minor dialplan snippets.


'dialplan show exten@context' also resolves various 'include='
directives.

If you had:

[local]
include = phones
exten = 120,1,Dial(SIP/trunk/123456)

[phones]
exten = 100,1,Dial(SIP/phone1)

the 'dialplan show local' would show the equivalent of

  include = phones
  exten = 120,1,Dial(SIP/trunk/123456)

whereas 'dialplan show 1...@local would show the actual (equivalent of)

  exten = 100,1,Dial(SIP/phone1)

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Re: [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?

2010-06-25 Thread Steve Edwards
On Fri, 25 Jun 2010, Gareth Blades wrote:

 If you are going to reboot the server regularly then make sure and 
 system updates are set to not automatically install new kernel versions. 
 Otherwise if you get a kernel update and reboot zaptel/dahdi wont load 
 until you recompile it.

I enhanced my /etc/init.d/zaptel to rebuild zaptel if the module was not 
available for the current kernel:

# check for a zaptel driver for this kernel
if  [ ! -s /lib/modules/`uname -r`/*/zaptel.ko ]
 thenecho Rebuilding zaptel
$0 rebuild
 fi

(and then a bit later in the file)

# rebuild the zaptel driver
 rebuild)
 if  [ -d /usr/src/zaptel/ ]
 thencd /usr/src/zaptel/
 make clean
 make
 make install
 elseecho You need to download and untar the zaptel
 echo driver source to /usr/src/zaptel.
 fi
 ;;

However, the philosophy of regularly rebooting the box ensures that you 
will never find the source of the problem.

Hey Gilles, any chance of you fixing whatever it is that you are doing 
that causes you to double-post EVERYTHING?

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

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Re: [asterisk-users] Big time system

2010-06-25 Thread David Backeberg
On Thu, Jun 24, 2010 at 11:24 PM, Cary Fitch ca...@usawide.net wrote:
 But, we have an opportunity to get into a big time telecom activity.

 It would have 2000 to 30,000 user lines per city, and we would like to have
 those brought back to a central location for control and because transport
 can be more economical than remote site rentals, maintenance and personnel.

I would say you need to make an RFP process to first negotiate your
calling rate extremely low with the major vendors of the country where
you're operating. If this is US, you're talking Qwest, ATT, Verizon,
and the ilk, and you negotiate an extremely low minute rate in return
for giving them a guaranteed minimum revenue. And while you're at it,
you ask them how they suggest you design the architecture over their
national network.

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Re: [asterisk-users] Big time system

2010-06-25 Thread Cary Fitch
Thanks for the feed back, but the rates are more or less predetermined.

ATT rates would be $.0007 per minute for local calls.  The operation would
be providing local phones wired to houses with copper pairs.

What I am looking for is the best ways to handle those lines when brought
to a local switch site.  The actual switch might not be there but back
hauled, might be a TDM switch, a concentrator (TNT, etc) 10 ganged
Asterisk systems, or tin can and string. 

I see some talking about TNTs in this forum.  Those are 672 lines or in some
versions double that, what is used behind them to do the processing, etc.

Cary Fitch

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: Friday, June 25, 2010 9:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Big time system

On Thu, Jun 24, 2010 at 11:24 PM, Cary Fitch ca...@usawide.net wrote:
 But, we have an opportunity to get into a big time telecom activity.

 It would have 2000 to 30,000 user lines per city, and we would like to
have
 those brought back to a central location for control and because transport
 can be more economical than remote site rentals, maintenance and
personnel.

I would say you need to make an RFP process to first negotiate your
calling rate extremely low with the major vendors of the country where
you're operating. If this is US, you're talking Qwest, ATT, Verizon,
and the ilk, and you negotiate an extremely low minute rate in return
for giving them a guaranteed minimum revenue. And while you're at it,
you ask them how they suggest you design the architecture over their
national network.

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[asterisk-users] Configure WAN Phone

2010-06-25 Thread Nicholas Hart
Hi,

I am relatively new to Asterisk and am looking for help in configuring an IP
based phone.  This phone is not on the same subnet as the PBX.  I read that
there could be an issue with NAT so I am bypassing this by connecting
temporarily with an Internet IP.  So far, I have configured the server
extension with the phone MAC address along with configuring the phone unit
with IP, Gateway and SIP info.  Upon rebooting the phone, I am getting 'NO
Service' message.   Kind of surprised there is no IP option on the PBX
Manager for this extension.  Any help would be much appreciated.  Thanks.

Info:
Asterisk v.  C.3.3.2
thirdlane PBX Manager
Aastra 6757i phone
Internet provider  -  Fios  (perhaps they are blocking ports?)

Best,
Nick
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Re: [asterisk-users] Configure WAN Phone

2010-06-25 Thread dotnetdub
On 25 June 2010 16:23, Nicholas Hart nh...@partsauthority.com wrote:


 Hi,

 I am relatively new to Asterisk and am looking for help in configuring an
 IP based phone.  This phone is not on the same subnet as the PBX.  I read
 that there could be an issue with NAT so I am bypassing this by connecting
 temporarily with an Internet IP.  So far, I have configured the server
 extension with the phone MAC address along with configuring the phone unit
 with IP, Gateway and SIP info.  Upon rebooting the phone, I am getting 'NO
 Service' message.   Kind of surprised there is no IP option on the PBX
 Manager for this extension.  Any help would be much appreciated.  Thanks.

 Info:
 Asterisk v.  C.3.3.2
 thirdlane PBX Manager
 Aastra 6757i phone
 Internet provider  -  Fios  (perhaps they are blocking ports?)

 Best,
 Nick


There is quite a bit of configuration to do to achieve what your trying to
do. You need to setup a TFTP server. The handset needs to know the address
of the TFTP server to download the config.
Head over to the thirdlane.com forums - they are a helpful bunch. Eric's
bark is worse than his bite :)


Brian
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Re: [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?

2010-06-25 Thread Tzafrir Cohen
On Fri, Jun 25, 2010 at 06:45:29AM -0700, Steve Edwards wrote:
 On Fri, 25 Jun 2010, Gareth Blades wrote:
 
  If you are going to reboot the server regularly then make sure and 
  system updates are set to not automatically install new kernel versions. 
  Otherwise if you get a kernel update and reboot zaptel/dahdi wont load 
  until you recompile it.
 
 I enhanced my /etc/init.d/zaptel to rebuild zaptel if the module was not 
 available for the current kernel:

You should look into dkms.

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Re: [asterisk-users] Big time system

2010-06-25 Thread Joe Freeman
Cary-

Asterisk may carry you a way down this road, but in the end, it's not, 
and was never designed to be a class 5 telecom switch. There are people 
working on a carrier grade implementation that may or may not be fully 
class 5, but I don't know what the status is on that. I haven't gotten 
an answer from Digium on that lately.

What you're looking for are local gateways that backhaul to a central 
switch site with equipment that can support traffic from multiple rate 
centers in multiple LATAs. This gets complicated quickly, especially if 
your rate centers are spread across multiple states.

You'll want some type of Multiservice Access Platform (MSAP). Zhone 
makes the MALC and their newer MXK box. Adtran has the TA-5000 shelf. 
Neither are what you'd call cheap. Both will provide T1 access, DSL, 
SDSL, VDSL, bonded, and even ethernet access to the customer over a 
variety of transport options, including copper pairs.

The Zhone box already has SIP backhaul for voice traffic, and the Adtran 
shelf should have it soon. Today the Adtran box has GR303 backhaul for 
voice.

All that said, what you're proposing indicates to me that you're likely 
to need to establish CLEC certification in whatever states you'll be 
operating. That in itself is not a short process. It can take anywhere 
from 90 days to a year depending on the state, and expect to spend from 
$10K up on legal costs per state alone. Insurance, financial health, and 
other requirements vary by state as well.

The ILECs generally won't even talk to you about establishing colo and 
gaining access to the copper loops until you get the CLEC certificate. 
Generally the process starts by getting the certificate, then 
negotiating an ICA, then trunking services, then colo. Different 
carriers will be easier to work with than others, but they are all a 
pain. ATT requires you to have a $10M general liability policy in place 
before you can even submit a request for a space availability report.

All this is not to say it can't be done, but to point out that it's a 
very difficult process to negotiate, even when you have done it several 
times. Without experience it can be close to impossible. I'd suggest 
getting a good telecom/clec consultant and a good telecom lawyer (I know 
a few) involved early in the process, or you'll end up spending ALOT of 
money.

Hit me off-list and I can give you more info.

Joe


On 6/24/2010 11:24 PM, Cary Fitch wrote:
 We are an asterisk user... small time system 50-100 users or so.

 But, we have an opportunity to get into a big time telecom activity.

 It would have 2000 to 30,000 user lines per city, and we would like to have
 those brought back to a central location for control and because transport
 can be more economical than remote site rentals, maintenance and personnel.

 We could take the local lines into concentrators (TNTs or equivalent) and
 bring back IP to a central site, or put servers at the remote cities.

 Our object is to serve as a central office switch for subscribers on
 standard telco service loops.

 This isn't a How many lines can I handle using a Belchfire 2600 processor?
 type question but a request for pointers to big time systems.  There would
 be no IP path to the end user, just copper.

 Thank you
 Cary Fitch




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Re: [asterisk-users] Big time system

2010-06-25 Thread David Backeberg
On Fri, Jun 25, 2010 at 11:00 AM, Cary Fitch ca...@usawide.net wrote:
 I see some talking about TNTs in this forum.  Those are 672 lines or in some
 versions double that, what is used behind them to do the processing, etc.

So a channelized DS3 is roughly 28*23 channels in US if you do one
D-channel per PRI (other options are possible). That gets you 644
channels. You can either buy gear that terminates a channelized DS3
natively, like a Cisco AS series device to voip-ify the PSTN channels,
or you can get a device like an Adtran MX2800 which breaks out the DS3
into individual T1/PRIs, which you can then terminate with a number of
different technologies, including a lot of Digium cards, or you can
voipify with appliances like a Cisco 3845.

So you can get a lot of asterisk boxes that have native DAHDI
channels, or you can put a layer in-between that adds expense, but
increases routing options.

That's how the DS3 works. To bundle DS3s, you generally get fiber to
the premise, and demux it at your data center using equipment approved
or provided by your telco of choice. If you're talking 30k channels,
that's some bigger glass, which then demuxes down to OC-whatever,
which eventually demuxes to lots of DS3s, but honestly I've never
worked at a scale past a handful of DS3s, so there may be a vastly
superior way to do things at that scale.

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Re: [asterisk-users] Big time system

2010-06-25 Thread Kevin P. Fleming
On 06/25/2010 10:00 AM, Cary Fitch wrote:
 Thanks for the feed back, but the rates are more or less predetermined.
 
 ATT rates would be $.0007 per minute for local calls.  The operation would
 be providing local phones wired to houses with copper pairs.
 
 What I am looking for is the best ways to handle those lines when brought
 to a local switch site.  The actual switch might not be there but back
 hauled, might be a TDM switch, a concentrator (TNT, etc) 10 ganged
 Asterisk systems, or tin can and string. 
 
 I see some talking about TNTs in this forum.  Those are 672 lines or in some
 versions double that, what is used behind them to do the processing, etc.

You really, really want to use IP backhaul as close to the end customers
as you can push it. If you can't, then you need to use multiplexing to
avoid having to have one channel per customer, which is excessive for
residential usage. This is what GR-303 was designed (and is still used) for.

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

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Re: [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?

2010-06-25 Thread Barry Miller
Hi Gilles.  You appear to be both posting to newsgroup
gmane.comp.telephony.pbx.asterisk.user AND sending the same message
directly to the asterisk-users list.  This means that we list subscribers
see two copies of all your messages: one from gmane, one from you.  (They
don't show up that way on gmane because it suppresses duplicates.)

Can you please pick one or the other?  Thanks.  Sorry for interrupting.

-- 
Barry

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[asterisk-users] Call drops on group paging asterisk - 1.4.22.1

2010-06-25 Thread das sandesh
Hi All,

We are using group paging and our asterisk version: 1.4.22.1, but when ever
any one page to the whole group (28 extensions), the calls which are on hold
on those extensions will be dropped, is there any other way to have this
feature or to go with Overhead paging. Currently this has become a serious
problem, can anyone through some light on this group paging senario?

Thank you very much

Regards
Sandesh
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[asterisk-users] Meetme delay - normal?

2010-06-25 Thread Mike
Hi,

 

I`ve done a few tests with comparing a two personne conf call (both phones
next to each other) and a simple call from one phone to the other.

 

In the simple call, there is virtually no delay when I am speaking. In the
meetme call, on an underutilized server, with a transcoder card for G729,
with one two participants, I have a relatively large delay (1/3 of a sec
maybe?).  There is an obvious difference in delay.

 

Is there anything I can do about that, or is this just something to live
with when using Meetme?

 

Using Asterisk 1.4.33.1, both Polycom phones using G729 in both cases.

 

Regards,

 

Mike

 

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Re: [asterisk-users] Call drops on group paging asterisk - 1.4.22.1

2010-06-25 Thread Mike
The phone brand and model might matter here, I have had no such problems
with Polycom phones.

 

Mike

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of das sandesh
Sent: Friday, June 25, 2010 12:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call drops on group paging asterisk - 1.4.22.1

 

Hi All,

We are using group paging and our asterisk version: 1.4.22.1, but when ever
any one page to the whole group (28 extensions), the calls which are on hold
on those extensions will be dropped, is there any other way to have this
feature or to go with Overhead paging. Currently this has become a serious
problem, can anyone through some light on this group paging senario?

Thank you very much

Regards
Sandesh

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Re: [asterisk-users] Big time system

2010-06-25 Thread Tarek Sawah

a Rack of load balanced Asterisk Servers with some customized billing system 
with a respectable centralized database like MsSQL or Oracle ..External E1 or 
T1 Gateways instead of TDM cards.. with load balancing?? as the whole operation 
is COPPER WEIRES .. can't that setup work for them?I'm asking as i'm looking 
for a similar setup just trying to set it up virtually before we go live.Regards

-- Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308






 Date: Fri, 25 Jun 2010 11:49:12 -0400
 From: j...@ngn-networks.com
 To: asterisk-users@lists.digium.com; ca...@usawide.net
 Subject: Re: [asterisk-users] Big time system
 
 Cary-
 
 Asterisk may carry you a way down this road, but in the end, it's not, 
 and was never designed to be a class 5 telecom switch. There are people 
 working on a carrier grade implementation that may or may not be fully 
 class 5, but I don't know what the status is on that. I haven't gotten 
 an answer from Digium on that lately.
 
 What you're looking for are local gateways that backhaul to a central 
 switch site with equipment that can support traffic from multiple rate 
 centers in multiple LATAs. This gets complicated quickly, especially if 
 your rate centers are spread across multiple states.
 
 You'll want some type of Multiservice Access Platform (MSAP). Zhone 
 makes the MALC and their newer MXK box. Adtran has the TA-5000 shelf. 
 Neither are what you'd call cheap. Both will provide T1 access, DSL, 
 SDSL, VDSL, bonded, and even ethernet access to the customer over a 
 variety of transport options, including copper pairs.
 
 The Zhone box already has SIP backhaul for voice traffic, and the Adtran 
 shelf should have it soon. Today the Adtran box has GR303 backhaul for 
 voice.
 
 All that said, what you're proposing indicates to me that you're likely 
 to need to establish CLEC certification in whatever states you'll be 
 operating. That in itself is not a short process. It can take anywhere 
 from 90 days to a year depending on the state, and expect to spend from 
 $10K up on legal costs per state alone. Insurance, financial health, and 
 other requirements vary by state as well.
 
 The ILECs generally won't even talk to you about establishing colo and 
 gaining access to the copper loops until you get the CLEC certificate. 
 Generally the process starts by getting the certificate, then 
 negotiating an ICA, then trunking services, then colo. Different 
 carriers will be easier to work with than others, but they are all a 
 pain. ATT requires you to have a $10M general liability policy in place 
 before you can even submit a request for a space availability report.
 
 All this is not to say it can't be done, but to point out that it's a 
 very difficult process to negotiate, even when you have done it several 
 times. Without experience it can be close to impossible. I'd suggest 
 getting a good telecom/clec consultant and a good telecom lawyer (I know 
 a few) involved early in the process, or you'll end up spending ALOT of 
 money.
 
 Hit me off-list and I can give you more info.
 
 Joe
 
 
 On 6/24/2010 11:24 PM, Cary Fitch wrote:
  We are an asterisk user... small time system 50-100 users or so.
 
  But, we have an opportunity to get into a big time telecom activity.
 
  It would have 2000 to 30,000 user lines per city, and we would like to have
  those brought back to a central location for control and because transport
  can be more economical than remote site rentals, maintenance and personnel.
 
  We could take the local lines into concentrators (TNTs or equivalent) and
  bring back IP to a central site, or put servers at the remote cities.
 
  Our object is to serve as a central office switch for subscribers on
  standard telco service loops.
 
  This isn't a How many lines can I handle using a Belchfire 2600 processor?
  type question but a request for pointers to big time systems.  There would
  be no IP path to the end user, just copper.
 
  Thank you
  Cary Fitch
 
 
 
 
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Re: [asterisk-users] Is there a default dial plan that is not in extention.conf?

2010-06-25 Thread Eyal Goltzman
Thank you all,

This is what I see after CLI dialplan show 1...@default :

  '100' =  hint: SIP/100IAX2/100
[pbx_config]
1. Dial(${HINT})
[pbx_config]
  '_1XX' = 1. Playback(digits/4)
[pbx_config]

From where come the 2 first lines?? I only have the third one as the only
one under my default context at extention.conf.

And what is [pbx_config]?

Thanks

Eyal

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Friday, June 25, 2010 4:05 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Is there a default dial plan that is not in
extention.conf?

On Fri, Jun 25, 2010 at 02:25:38PM +0300, Eyal Goltzman wrote:
 Hi,
 
  
 
 I have a trivial peace of dialplan for exten 100. I try to change it to
_1XX
 and the asterisk act according to a different (Default??) dial plan and
not
 the one I want? Is that possible? Where is the other dialplan sits? In my
 extention.conf I can't see something that look like what asterisk is
 dialing.
 
 How can I trace\debug my dialplan?

To see where it comes from, run in the Asterisk CLI:

  dialplan show context

or:

  dialplan show exten@context

Here is a partial output from 'dialplan show' here, that shows all of
them (but is normally overly long)

[ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ]
  's' =1. NoOp() [app_queue]

[ Context 'parkedcalls' created by 'features' ]
  '700' =  1. Park() [features]

[ Context 'app_dial_gosub_virtual_context' created by 'app_dial' ]
  's' =1. NoOp() [app_dial]

[ Context 'from-pstn' created by 'pbx_config' ]
  '_X.' =  1. Answer()   [pbx_config]
2. Playback(demo-instruct)[pbx_config]
3. Hangup()   [pbx_config]

[ Context 'ael-dundi-e164' created by 'pbx_ael' ]
  's' =1. MSet(LOCAL(exten)=${ARG1}) [pbx_ael]
2. Goto(${exten},1)   [pbx_ael]
3. Return()   [pbx_ael]


'pbx_config' is dialplan that was generated from your extensions.conf. 
'pbx_ael' is dialplan that was generated from extensions.ael.
Various other modules include their own minor dialplan snippets.


'dialplan show exten@context' also resolves various 'include='
directives.

If you had:

[local]
include = phones
exten = 120,1,Dial(SIP/trunk/123456)

[phones]
exten = 100,1,Dial(SIP/phone1)

the 'dialplan show local' would show the equivalent of

  include = phones
  exten = 120,1,Dial(SIP/trunk/123456)

whereas 'dialplan show 1...@local would show the actual (equivalent of)

  exten = 100,1,Dial(SIP/phone1)

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

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[asterisk-users] sip_xmit: sip_xmit returned -1: Operation not permitted

2010-06-25 Thread Jonas Kellens

Hello,

my Asterisk CLI is flooded with the following message :

[Jun 25 21:24:57] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit 
of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation 
not permitted
[Jun 25 21:25:01] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit 
of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation 
not permitted
[Jun 25 21:25:05] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit 
of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation 
not permitted
[Jun 25 21:25:09] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit 
of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation 
not permitted
[Jun 25 21:25:13] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit 
of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation 
not permitted



I have no idea where this IP comes from, there is no SIP peer or user 
with this IP-address.


What can I do to get ride of this message that is constantly flooding my 
CLI ?!



Reloading or restarting my Asterisk does not help !


Kind regards,

Jonas.
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Re: [asterisk-users] UDPTL T38 via NAT

2010-06-25 Thread Martin
 I've got the following setup :
 [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP]
I don't see where your NAT is in this scenario

 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29)

 This means my outgoing udptl traffic is correctly translated, but
 somehow i'm sending 172.16.0.156 instead of my public IP address on the
 firewall.
What about externip=62.180.xxx.xxx?

 Did you try t38pt_usertpsource=yes ?
AFAIK this is about a port used for rtp, not ip address...

I'm currently trying this over 2 NATs against eachother (yes, the worst 
case) with some ports forwarded but with rare success. One of those NAT's 
rewrites a port numbers for some reason (i see the ATA registered on port 50xxx 
or so, the same for rtp. I think t38pt_usertpsource is meant for such a case...?
[asterisk 1.6]-LAN-[NAT gateway]inet-[NAT gateway]-LAN-[ATA]-[FAX]
Has anybody some positive experience with this?
Any idea why NAT messes up the port numbers?
Martin L 


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Re: [asterisk-users] Local channel usage

2010-06-25 Thread Steve Edwards
On Tue, 22 Jun 2010, Philipp von Klitzing wrote:

 Here's an example for Voicemail live that uses such a technique: 
 http://www.voip-info.org/wiki/view/Asterisk+tips+voicemail+live

A great example. If I ever get around to setting up a home Asterisk 
server, I'm sure this will help with the WAF.

It looks like you may have a couple of bugs in your AGIs:

//read the standard agi variables
 while (!feof($in)) {
$temp = str_replace(\n,,fgets($in,4096));
$s = split(:,$temp);
$agi[str_replace(agi_,,$s[0])] = trim($s[1]);
if (($temp == ) || ($temp == \n)) {
break;
}
 }

The tests for $temp being empty should be before the split. If you execute 
the script from the command line, the PHP interpreter will complain about 
$s[1].

Also in your __read__() and __write__() functions:

function __write__($line) {
   global $debug;
   if ($debug) echo VERBOSE \write: $line\\n;
   print $line.\n;
}

If $debug is set, you issue the agi VERBOSE request but you do not read 
the response. This violates the AGI protocol and leaves the response 
hanging in STDIN where it is subsequently read by the GET VARIABLE 
request.

-- 
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] Non-native codecs - MELPe?

2010-06-25 Thread Kirin, Carol (IS)

Has anyone needed a coded that Asterisk does not natively support, such
as MELPe or CVSD? If so, did you find a pure software solution and
provided that as an addition to Asterisk? Was that solution successful?
Has using an I/F card with a DSP proved to be the better solutions? We
are beginners with Asterisk so any help/advice on how to best implement
non-native Codecs into Asterisk will be welcome.

CK

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Re: [asterisk-users] Is there a default dial plan that is not in extention.conf?

2010-06-25 Thread Tzafrir Cohen
On Fri, Jun 25, 2010 at 10:19:01PM +0300, Eyal Goltzman wrote:
 Thank you all,
 
 This is what I see after CLI dialplan show 1...@default :
 
   '100' =  hint: SIP/100IAX2/100
 [pbx_config]
 1. Dial(${HINT})
 [pbx_config]
   '_1XX' = 1. Playback(digits/4)
 [pbx_config]
 
 From where come the 2 first lines?? I only have the third one as the only
 one under my default context at extention.conf.
 
 And what is [pbx_config]?

extentions.conf (technically: read by the module pbx_config.so).

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

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