Re: [asterisk-users] Replacing digital pri card

2011-01-03 Thread Matt Watson
I don't imagine this would be too complicated - don't have any experience
with AsteriskNOW - but on a 'vanilla' linux distro it would just be a matter
of making sure dahdi is loading the correct drivers and doing a couple of
minor config file updates.



On Tue, Dec 28, 2010 at 3:01 PM, Tyler Davis tda...@zulily.com wrote:

 I need to replace our current 1 port pri card with a quad port card. I'm
 currently using the newest AsteriskNOW distro. Are there any issues I should
 expect to run into? I'm hoping the transition will be smooth, however I
 havent had to do this in the past.

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Re: [asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??

2010-11-30 Thread Matt Watson
Just out of curiosity, what country are you in?

I agree with the others in this thread, this seems very bizzare that the
telco requires you to do SS7 for dialup connections.  I would ask them for
specifics about the legal issues with what you are doing - it sounds to me
like they are just trying to upsell you on a more expensive product.

I am in Canada and we run exactly the configuration you are currently
doing... we still have dialup internet customers that dial into AS5300's via
PRI's.  Our telco has a PRI product gear specifically for this use... they
call it 'ISP-PRI' I'm not entirely sure what the restriction is on it I
have also just kind of assumed that it is inbound calls only, but I've never
tried making outbound calls on them.  I do know they 25-30% cheaper than our
regular voice PRIs though.

--
Matt


2010/11/24 José Pablo Méndez Soto aux...@gmail.com

 Hello,

 We are working on implementing a solution for a medium service provider.
 They were previously using a Cisco AS5300 gateway with some PRI trunks to
 receive modem calls, then route them out the Internet.

 The Telco they were buying the trunks to discovered this configuration and
 restricted them due to legal conventions, and stated that in order to
 continue doing this, they would have to talk SS7 directly.


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Re: [asterisk-users] OT: for those wondering on the stability

2010-11-30 Thread Matt Watson
I'd suggest you take the opportunity to patch that box while its going to go
offline anyways!

*nix is not immune to security problems despite what some might like to
think!  Also, I believe Asterisk has had a couple relatively
serious vulnerabilities that have been patched in 1.4 and 1.6 over the last
2 years - of course you could of already patched those without a reboot.



On Mon, Nov 29, 2010 at 11:13 PM, C F shma...@gmail.com wrote:

 r...@pbx:~# uptime
  23:10:15 up 606 days,  9:38,  1 user,  load average: 0.31, 0.08, 0.02

 Customer called they are having a scheduled power outage for most of
 the day because of construction if I can shut down the machine
 gracefully. So I decided to run uptime first.
 Enjoy

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Re: [asterisk-users] Asterisk T38

2010-09-23 Thread Matt Watson
On Wed, Sep 22, 2010 at 10:46 AM, Adam Moffett a...@plexicomm.net wrote:

 That's probably what I'm going to have to do.  Thanks.

  I suppose that merely removing ATA and asterisk from the middle, and
  plugging a pots line into a fax machine is out of the question.
 
 



If you are running 1.6, you can try this -
https://issues.asterisk.org/view.php?id=13405  I have this implemented on
our production system but I personally get hit and miss results with it.
 Sometimes it doesn;t seem to detect its a fax call.   The fax typically
goes through fine over g.711u though as long as you have good network links.
 My setup previously included a couple of lan-extensions over DSL and it was
pretty unreliable on them if there was other network traffic going on even
with QoS on the edge routers heavily favoring the phone traffic.

https://issues.asterisk.org/view.php?id=13405If you are running 1.4 you
can try this - http://www.zoiper.com/foip/  I used this back before
zoiper acquired the company that owned it and it worked absolutely perfect.
 I haven;t tried it since its been open sourced and had 3rd party people
patch it current versions of asterisk though.
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Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

2010-07-06 Thread Matt Watson
On Tue, Jun 29, 2010 at 10:39 AM, William Stillwell (Lists) 
william.stillwell-li...@ablebody.net wrote:

 I use SecureCRT+FX , and use ansi graphics.

 Putty is nice w/WinSCP as well.




I'll +1 this - SecureCRT+FX is the first thing I got my employer to buy a
license of for me when I had to start using Windows on my desktop for other
reasons instead of a Linux Distro.  I do keep a copy of PuTTY handy on too
though, the great thing about putty is that it doesn;t require it to be
installed on a desktop, so you can just keep a copy of the executable on a
USB flash drive or Windows share that you can then run from any desktop if
you happen to be at a computer other than your own.

PuTTY works well, but there are some things in it that just drive me
absolutely crazy, like right-click in the window is an automatic paste... i
have pasted into a putty window by accident more times than I can count,
most of the time its from doing a right-click expecting a context menu to
get a 'copy' action and then i just end up pasting what i actually wanted to
copy :|

--
Matt

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Re: [asterisk-users] How are your PRI interrupts balanced? (+ Soft lockup BUG)

2010-03-29 Thread Matt Watson
Dell server by any chance?

I have a similar problem with a TE220B in a Dell 1950 III server - i've seen
several other people having issues with digium cards in dell servers as
well.

I've actually done something similar to what you have done - isolated the
TE220B onto its own IRQ and set processor affinity for all the IRQs to
particular cores... so far I haven't had kernel pancs since doing this, but
its still a little too early to say if it has fixed the issue 100% or not.

--
Matt

On Mon, Mar 29, 2010 at 8:30 PM, James Lamanna jlama...@gmail.com wrote:

 Hi,
 I'm trying to figure out the cause of a soft lockup I experienced:

 Mar 29 09:38:24 pstn1 kernel: BUG: soft lockup - CPU#0 stuck for 10s!
 [asterisk:32029]
 Mar 29 09:38:24 pstn1 kernel: Pid: 32029, comm: asterisk
 Mar 29 09:38:24 pstn1 kernel: EIP: 0060:[c046e7fe] CPU: 0
 Mar 29 09:38:24 pstn1 kernel: EIP is at kfree+0x68/0x6c
 Mar 29 09:38:24 pstn1 kernel:  EFLAGS: 0286Tainted: GF
 (2.6.18-128.1.10.el5 #1)
 Mar 29 09:38:24 pstn1 kernel: EAX: 0029 EBX: f7ff9380 ECX:
 f7fff880 EDX: c11ff9a0
 Mar 29 09:38:24 pstn1 kernel: ESI: 0286 EDI: cffcda00 EBP:
 e5e10c80 DS: 007b ES: 007b
 Mar 29 09:38:24 pstn1 kernel: CR0: 80050033 CR2: b7ce39e0 CR3:
 0f911000 CR4: 06d0
 Mar 29 09:38:24 pstn1 kernel:  [c05b067c] kfree_skbmem+0x8/0x61
 Mar 29 09:38:24 pstn1 kernel:  [c05e9aaf] __udp_queue_rcv_skb+0x4a/0x51
 Mar 29 09:38:24 pstn1 kernel:  [c05ad993] release_sock+0x44/0x91
 Mar 29 09:38:24 pstn1 kernel:  [c05ea939] udp_sendmsg+0x44e/0x514
 Mar 29 09:38:24 pstn1 kernel:  [c05efdec] inet_sendmsg+0x35/0x3f
 Mar 29 09:38:24 pstn1 kernel:  [c05ab30c] sock_sendmsg+0xce/0xe8
 Mar 29 09:38:24 pstn1 kernel:  [c043464f]
 autoremove_wake_function+0x0/0x2d
 Mar 29 09:38:24 pstn1 kernel:  [c04ea17b] copy_from_user+0x17/0x5d
 Mar 29 09:38:24 pstn1 kernel:  [c04ea3a1] copy_to_user+0x31/0x48
 Mar 29 09:38:24 pstn1 kernel:  [f89ab141] zt_chan_read+0x1e0/0x20b
 [zaptel]
 Mar 29 09:38:24 pstn1 kernel:  [c04ea195] copy_from_user+0x31/0x5d
 Mar 29 09:38:24 pstn1 kernel:  [c05ac4c4] sys_sendto+0x116/0x140
 Mar 29 09:38:24 pstn1 kernel:  [c0415d4f] flush_tlb_page+0x74/0x77
 Mar 29 09:38:24 pstn1 kernel:  [c0461331] do_wp_page+0x3bf/0x40a
 Mar 29 09:38:24 pstn1 kernel:  [c04284f1] current_fs_time+0x4a/0x55
 Mar 29 09:38:24 pstn1 kernel:  [c0488f9b] touch_atime+0x60/0x91
 Mar 29 09:38:24 pstn1 kernel:  [c047d9d0] pipe_readv+0x315/0x321
 Mar 29 09:38:24 pstn1 kernel:  [c05acde4] sys_socketcall+0x106/0x19e
 Mar 29 09:38:24 pstn1 kernel:  [c0404f17] syscall_call+0x7/0xb
 Mar 29 09:38:24 pstn1 kernel:  ===


 This occurred during a high load period (52 calls across 3 PRI spans).

 A couple days ago I moved the interrupts for my PRI card to CPU0 from
 CPU3, because CPU3 was handling everything else:
   CPU0   CPU1   CPU2   CPU3
  0:306  0  0 3684057379IO-APIC-edge  timer
  1:  0  0  0  13468IO-APIC-edge  i8042
  8:  0  0  0  3IO-APIC-edge  rtc
  9:  0  0  0  0   IO-APIC-level  acpi
  12:  0  0  0  4IO-APIC-edge  i8042
 169:  0  0  0  0   IO-APIC-level
  uhci_hcd:usb2
 177:  0  0  0   18392593   IO-APIC-level  ata_piix
 185:  0  0  0  1   IO-APIC-level
  ehci_hcd:usb1
 193:  0  0  0  0   IO-APIC-level
  uhci_hcd:usb3
 201:  0  0  0 2090021759   IO-APIC-level  eth0
 209:  149621223  0  0 3534419461   IO-APIC-level  wct4xxp


 (The CPU3 number for wct4xxp is not increasing any more).

 What is the interrupt distribution of other people's systems?
 Before I made this change I was having a problem with D-channels
 dropping occasionally, so I thought it might be an interrupt/load
 issue.

 Thank you.

 -- James

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Re: [asterisk-users] Aastra weirds IP 169.x.x.x

2010-03-24 Thread Matt Watson
You should be safe to upgrade to the latest Aastra firmware - i've never had
problems with any of the Aastra upgrades - that being said, thats probably
better asked to Aastra support instead of here.

As for the 169.254.0.0/16 address...

theres nothing weird about these addresses... 169.254.0.0/16 is a block
designated as the link-local address block, operating systems can assign an
IP in this range to themselves automatically when there is either
no statically configured address or they are unable to contact a DHCP
server.  Its the same with the FE80::/10 block in IPv6.  Nearly all of the
time you see these addresses its a problem where your devices are unable to
talk to your DHCP server for some reason.

I'm actually a little surprised that you are able to have those phones
connect to your Asterisk server at all... do you have a
169.254.0.0/16address assigned on your server or a router between the
phones and Asterisk
server?

--
Matt

On Wed, Mar 24, 2010 at 7:07 PM, Danny Dias ing.diasda...@gmail.com wrote:

 Hello my friends...

 Currently we are using the following firmware versions on ours aastra 55i:

 Firmware Information
 Attribute Value
 Firmware Version 2.1.0.2145
 Firmware Release Code SIP
 Boot Version 2.0.1.1055
 Date/Time Jun 20 2007 06:20:29

 Can we make a firmware upgrade to the latest one: 6755i (55i) SIP,
 V2.5.3.18, January 2010 , English , ZIP , 2,849 KB

 on the site:
 http://www.aastra.com/cps/rde/xchg/SID-3D8CCB6A-FE6670DF/04/hs.xsl/19705.htm

 Could this make problems?

 We are receving very weirds ip on the server, see here:

 mypbx*CLI sip show channels
 Peer User/ANRCall ID  Seq (Tx/Rx)  Format
 Hold Last Message
 10.4.1.130   (None)  b7dd744679a  00101/14365  0x0 (nothing)No
   Rx: REGISTER
 10.4.1.151   308 8cbe459c33e  00102/16076  0x0 (nothing)No
   Tx: NOTIFY
 10.4.1.144   368 607d5af86cd  00102/25625  0x0 (nothing)No
   Tx: NOTIFY
 169.254.236.26   308 4f407ce65eb  00102/15097  0x0 (nothing)No
   Tx: NOTIFY
 169.254.21.164   309 f1e31e48e10  00102/23948  0x0 (nothing)No
   Tx: NOTIFY
 5 active SIP channels

 Can you see the 169.x.x.x? whats the meaning of this? is this a serious
 firmware problem? dhcp problem?

 Thanks in advance

 DD

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Re: [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.

2010-03-17 Thread Matt Watson
On Tue, Mar 9, 2010 at 6:31 PM, Klaus Darilion klaus.mailingli...@pernau.at
 wrote:



 Attached is an untested (I did not had the time yet) port to Asterisk
 1.4.29.1 (DAHDI). Maybe the modules need some adaptions too.

 Maybe someone wants to give it a try.

 regards
 klaus


Just as an FYI, your 1.4.29.1 patch applies successfully against 1.4.30 as
well.  I've got a patched 1.4.30 system compiled and ready to install later
tonight during off-hours and will begin having people test tomorrow.  People
here are going to be quite thrilled about having T.38 transparent gatewaying
again.

--
Matt
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Re: [asterisk-users] dnd

2010-03-16 Thread Matt Watson
On Mon, Mar 15, 2010 at 1:44 PM, Ott Rose sixfourimp...@hotmail.com wrote:

  I did a clean install to freepbx 2.6.1 and now when i do *76 i get a 1
 second flash on the screen then the phone hangs up. the FOP says it is on
 DND but some ext are still getting calls. once i do a *76 FOP still says I
 am on dnd. I am running asterisk 1.6.0.21.

 before i was getting a message like dnd activated and dnd deactivated.


 i posted this on the freepbx site and here is what i got


 *DND should not have seen any change in 2.6.1???*



I sympathize with your problem, but posting a fpbx-specific problem here is
kind of like calling up Microsoft and asking Why does Adobe Photoshop  say
Error 652 when I click on the eraser.

You may want to try messaging Phillipe, one of the fpbx devs, on the fpbx
forums, he is normally pretty helpful.  His username on their forums is
p_lindheimer.

The first thing you probably want to do however, is turn up verbose and
perhaps debug logging on Asterisk to find out exactly what is going on when
you are trying to turn on/off DND.  As for not hearing anything, it may be
something as simple as audio files being missing or in the wrong location.
 That may account for you not getting audio feedback when turning dnd
on/off.

You may not get everything you need for trying to figure out why calls are
coming through on extensions with DND turned on from Asterisk logging though
as fbpx does a lot of call-setup stuff through an AGI file -
'dialparties.agi' and I'm not sure how good/bad its logging is.

Have you considered simply upgrading to the latest fpbx release?  2.7 was
released about a week or 2 ago.

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Re: [asterisk-users] MWI and 1.6.1

2010-03-12 Thread Matt Watson
Hi Dave,

Thought I'd give you an update - I completely rebuilt my astdb the other
night by renaming it, having * recreate it and then re-creating all my
custom entries in it.

Didn't have any effect, I had somebody report false MWI notifications again
earlier this morning.

--
Matt

On Tue, Mar 9, 2010 at 11:34 PM, Matt Watson m...@mattgwatson.ca wrote:

 Hi Dave,

 Sure enough my astdb does contain references to VM files as shown with
 strings - doing the database dump however does not show the references.

 I'm not sure about the internals of how Berk DB works, however I;m also
 seeing references to lots of other data that really shouldn't be part of my
 config anymore either - like I can see some employee names that are no
 longer a part of our company and thus have been deleted from our * config,
 some several years ago.  I suspect that berkdb is just not overwriting some
 of the data for whatever reason and has some internal mechanism for knowing
 what to ignore.

 I believe I can probably test your theory tomorrow evening though, I don't
 think I have too much in my astdb that can't be easily re-created, I think I
 can probably delete my astdb entirely and regenerate it.  I'll just need to
 take a closer look at it first though.

 I would however like to believe that if * is no longer supposed to be using
 berkdb for any VM reference data, that any calls to read the voicemail
 counts from the DB should have been removed.


 --
 Matt

 On Mon, Mar 8, 2010 at 5:08 PM, Dave Poirier davepoir...@gmail.comwrote:


 So a couple of questions I have for you Matt...
 If you run strings on your astdb file are you seeing references to
 messages files in it?

 #strings /var/lib/asterisk/astdb | grep -i msg

  and if so...

 If you run a db_dump185 on your astdb file do the references go away?

 #db_dump185 -p -f /tmp/astdb.dump astdb


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Re: [asterisk-users] MWI and 1.6.1

2010-03-09 Thread Matt Watson
Hi Dave,

Sure enough my astdb does contain references to VM files as shown with
strings - doing the database dump however does not show the references.

I'm not sure about the internals of how Berk DB works, however I;m also
seeing references to lots of other data that really shouldn't be part of my
config anymore either - like I can see some employee names that are no
longer a part of our company and thus have been deleted from our * config,
some several years ago.  I suspect that berkdb is just not overwriting some
of the data for whatever reason and has some internal mechanism for knowing
what to ignore.

I believe I can probably test your theory tomorrow evening though, I don't
think I have too much in my astdb that can't be easily re-created, I think I
can probably delete my astdb entirely and regenerate it.  I'll just need to
take a closer look at it first though.

I would however like to believe that if * is no longer supposed to be using
berkdb for any VM reference data, that any calls to read the voicemail
counts from the DB should have been removed.


--
Matt

On Mon, Mar 8, 2010 at 5:08 PM, Dave Poirier davepoir...@gmail.com wrote:


 So a couple of questions I have for you Matt...
 If you run strings on your astdb file are you seeing references to messages
 files in it?

 #strings /var/lib/asterisk/astdb | grep -i msg

  and if so...

 If you run a db_dump185 on your astdb file do the references go away?

 #db_dump185 -p -f /tmp/astdb.dump astdb


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Re: [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.

2010-03-08 Thread Matt Watson
Awesome!

I was an Attrafax customer and was very disappointed when it vanished and
couldn;t get new modules for newer versions Asterisk with our paid license.

If anybody is working on t38 gatewaying code for 1.6, it would be worth a
look at this, as I can attest that Attrafax worked quite well at t38
gatewaying.

--
Matt

On Sun, Mar 7, 2010 at 4:52 AM, Zoa zoach...@securax.org wrote:


 On friday we finally released Attrafax under a GPL2 license.
 It comes with its own set of modems and built in transparent gatewaying.
 The solution should be quite stable as long as the line quality is ok.
 (Some tools for measuring the line quality are included in the release,
 as well as some fax2mail scripts).

 There is an example implementation included for Asterisk 1.4, if someone
 wants to porting it to the new fax backend or more recent asterisk
 versions and needs some help, let us know.

 The full press release can be found here:
 http://www.zoiper.com/downloads/opensource_fax_stack_PR.pdf the project
 homepage can be found at www.zoiper.com/foip/

 Cheers,

 Zoa

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Re: [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.

2010-03-08 Thread Matt Watson
I just downloaded a copy of this, by any chances does Zoiper by any chance
have diff files available for a more recent 1.4.x release?  (I know 1.6 is
probably out of the question)

Thanks,

--
Matt

On Mon, Mar 8, 2010 at 12:11 PM, Matt Watson m...@mattgwatson.ca wrote:

 Awesome!

 I was an Attrafax customer and was very disappointed when it vanished and
 couldn;t get new modules for newer versions Asterisk with our paid license.

 If anybody is working on t38 gatewaying code for 1.6, it would be worth a
 look at this, as I can attest that Attrafax worked quite well at t38
 gatewaying.

 --
 Matt

 On Sun, Mar 7, 2010 at 4:52 AM, Zoa zoach...@securax.org wrote:


 On friday we finally released Attrafax under a GPL2 license.
 It comes with its own set of modems and built in transparent gatewaying.
 The solution should be quite stable as long as the line quality is ok.
 (Some tools for measuring the line quality are included in the release,
 as well as some fax2mail scripts).

 There is an example implementation included for Asterisk 1.4, if someone
 wants to porting it to the new fax backend or more recent asterisk
 versions and needs some help, let us know.

 The full press release can be found here:
 http://www.zoiper.com/downloads/opensource_fax_stack_PR.pdf the project
 homepage can be found at www.zoiper.com/foip/

 Cheers,

 Zoa

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Re: [asterisk-users] MWI and 1.6.1

2010-03-04 Thread Matt Watson
I'm having this EXACT same problem, I haven;t been able to narrow down the
cause of it yet, but it seems to me that users are receiving notifications
for voicemails in mailboxes that belong to other people, as sometimes their
mail count magically disappears, which I have been suspecting is when
somebody else checks their VM.

I found the problem also exists in 1.6.2 which is where I first noticed it
(upgraded from 1.4.x to 1.6.2.x).  I tried downgrading to 1.6.1 and the
problem seemed not quite as bad, but I know its still present.  I was
actually quite surprised to find that nobody had previously mentioned the
problem on this list when I came across it so I thought it might of been
something specific to my situation.

Even if you turn the polling options back on in the voicemail conf file the
problem still persists.

We are using all Aastra phones - a mix of 9133i, 9112i, 480, 35i, 57i phones
- but the problem seem unrelated to the make/model of the phone based on
seeing you having the same problem with Polycom's.

Not sure that it should matter, but we are using FreePBX 2.6 ontop of
asterisk and running it in users and devices mode (as apposed to the
default extensions mode).

If you do a voicemail show users from the Asterisk console it shows the
correct VM counts for the mailboxes, so its not that Asterisk is counting
them incorrectly, it just seems to be sending the notifications of VMs to
the wrong places.

I'm suddenly very glad I;m not alone on this one!

I;m more than happy to do any testing of patches if anybody has any
suggestions.

--
Matt


On Tue, Mar 2, 2010 at 1:36 PM, Dave Poirier dpoir...@mesd.k12.or.uswrote:

 We are having an issue with Asterisk 1.6.1 and the MWI turning on when a
 user doesn't have voicemail. We see random MWI lights come on and the phone
 indicates a random number of messages (its been anywhere from 1-14) when a
 server reload is done.

 I just checked one user, they have no messages old or new and the phone
 (Polycom IP330) indicates that they have 2 messages. The user will check for
 messages, the system will tell them that they have none and the light goes
 out.

 I know that starting in 1.6 Asterisk moved from a polling system to an
 event based system but it's unclear to me what is causing these events to be
 generated. Anyone else experience this? Any tips, suggestions?

 Thanks,
 Dave


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Re: [asterisk-users] Parsing Asterisk's .conf files from Perl, Java or PHP file

2009-05-22 Thread Matt Watson
There already is a special character to tell asterisk not to parse a line...
its: ;  that is why the default configuration is completely filled with
lines that start with ;  its considered a comment character to asterisk and
will make it ignore the line... you'd just want to add some extra characters
to your program probably... something that would denote this is a config
line for script A

so if you make your lines something like


;A:

then you could have another character for script B that would have lines in
the same file like:

;B:

or w/e... its up to you to determine what characters you'd want to use
beyond the ; to denote config lines for your other programs.

--
Matt

On Fri, May 22, 2009 at 3:23 AM, Olivier oza-4...@myamail.com wrote:

 Hi,

 To a large extend, Asterisk's /etc/asterisk/*.conf configuration files
 conform to a format such as:

 [section1]
 key1=value1
 key2=value2

 [section2]
 key1=value1
 key2=value2
 ...

 To increase coherence when running custom-made application in Perl, Java,
 PHP, ...) and Asterisk on the same platform, I'm wondering if could extend a
 bit Asterisk's config files instead of duplicating data in an independant
 config file.

 For instance, an app that uses Manager interface needs to be configured
 with :
 - the Asterisk manager interface IP address,
 - a username and secret.

 The later 2 parameters are included /etc/asterisk/manager.conf but the
 first one is not.
 So instead of writing an independant myapp.conf holding all 3 parameters,
 should I only add the first parameter to existing manager.conf file ?

 Doing this, I would have to make sure that when Asterisk is parsing its
 config file, it doesn't stop when it reads unkown supplementary parameters
 (those added for custom app).
 It seems to be the case now with 1.6.1 : a NOTICE warning is sent but it
 doesn't really hurt.
 [May 22 09:15:32] NOTICE[15917]: manager.c:3903 __init_manager: Invalid
 keyword foo = bar in manager.conf [general]

 Maybe, adding a keyword that would tell Asterisk to skip reading this
 config file line would be a plus (avoiding warnings and collision with new
 keywords).
 What do you think ?

 Then, my next question, is there widely available librairies to parse
 Asterisk's config files-like files ?

 Regards


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Re: [asterisk-users] Voicemail and remote directory with SSHFS

2009-05-21 Thread Matt Watson
Not that I;m exactly a big fan of NFS but... why would you choose to
implement a filesystem that was designed to emulate Windows shares for your
UNIX-type environment?  You have to kind of expect odd problems like this
when you choose to use things for other than their intended purpose.  Samba
I would say is probably alot more focused on providing storage shares for
Windows desktop clients, not for UNIX-type clients.  Sure there is some
support to do what you want, but just keep in mind that similiar to using
sshfs like you were trying before, Samba, was really not designed to be used
by UNIX clients.  You've already found the most obvious reason... case
sensative filenames - which Windows does not support, and UNIX programs
expect filesystems on your UNIX machine *will* support it.

That seems kind of like me deciding to use ntfs on a local partition on
linux box instead of ext3/4, jfs, reiserfs, etc.

--
Matt

On Thu, May 21, 2009 at 5:06 AM, Elliot Murdock murdo...@gmail.com wrote:

 Hello!

 Thanks...I set up a Samba mount, which works ok, except that Asterisk
 confuses a wave file as a wav49 file.  I think it may have something do with
 the way Samba supports case sensitivity.  Since Windows is not very
 aggressive when it comes to being case sensitive, I am thinking that Samba
 is saving files with the last three characters, wav, as uppercase, WAV.

 What is the procedure to ensure all the files are saved as is in Samba?

 Thanks,
 Elliot


 On Thu, May 14, 2009 at 5:12 PM, Tilghman Lesher 
 tilgh...@mail.jeffandtilghman.com wrote:

 On Thursday 14 May 2009 08:14:17 Elliot Murdock wrote:
  The problem is a file locking problem that Asterisk needs to make
 changes
  to the directory.  I was initially shying away from NFS and Samba,
 because
  I prefer to avoid any sort of security issues with only remotely
 mounting
  one or two directories.  NFS and Samba are designed for larger
  applications, which makes those types of technology worthwhile.

 No, they're both designed as filesystems, which makes typical things like
 locking possible.  SSH is designed as a communications medium, and someone
 has hacked filesystem support on top of it (poorly, apparently).  SSHFS
 was
 never designed to be used in server production environments and should not
 be used there.

  I am wondering if there is any way to disable Asterisk's request to lock
  the directory.  I know this may cause some loss in data, but for the
 volume
  voicemail receives, it should be rare enough that would make this
 approach
  an option.

 There is not.  Use a real filesystem that supports file locking (or
 really,
 file linking, which is how the locking is implemented) procedures.

 --
 Tilghman

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Re: [asterisk-users] From 1.4 to 1.6.0

2009-05-21 Thread Matt Watson
I'd be interested in this as well... I;m coming up to an upgrade cycle and
trying to decide if I should upgrade to the latest 1.4 or 1.6.1

When others that have commented on this say they have had problems with PSTN
connections, are you referring to T1 or POTS?  I;m in a T1 scenerio, so if
problems are specific to POTS then thats obviously not a deal breaker for my
setup.

Thanks,

--
Matt

On Thu, May 21, 2009 at 10:02 AM, Danny Nicholas da...@debsinc.com wrote:

  My issues are all DAHDI/POTS related.  Unfortunately, our present
 communication depends on the POTS lines, so I’m back to 1.4.25-rc1 as stated
 earlier.


  --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
 *Sent:* Thursday, May 21, 2009 8:53 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] From 1.4 to 1.6.0





 2009/5/18 Danny Nicholas da...@debsinc.com

 I'd love to see this as well.  After a few days of trying 1.6.1 (from
 1.4.21) I dropped back to 1.4.25-rc1 and that is going pretty well.



 Which issues did you get ?
 I'm about to deply a 1.6.1 system it does seem to work ok in a pure SIP
 environment.




 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel
 Molina
 Sent: Monday, May 18, 2009 3:31 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] From 1.4 to 1.6.0

 Hi everyone,

 I was just wondering, does anyone managing production asterisk servers
 migrated successfully from 1.4.2X to 1.6.0.X? I would like to see if
 there are some successful cases. Is your 1.6.0.X behaving well, with
 acceptable stability? Please share your experiences.

 Thanks,

 --
 Ing. Miguel Molina
 Grupo de Tecnología
 Millenium Phone Center
 PBX: (+57 1)6500800 ext. 1201
 Fax: (+57 1)6500816
 Móvil: (+57)3138873587


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Re: [asterisk-users] Aastra 9133i programmable buttons (* 4.1.23)

2009-03-17 Thread Matt Watson
The 1.x firmware for Aastra's (for the 9112i / 9133i / 480i) do support some
of the XML functionality that you see in the newer 2.x firmware (for the
more recent models).

I;m not sure if controlling LED status of the keys is supported by 1.x - but
you should be able to find that out by taking a look at Aastra's XML API
document here:
http://www.aastra.com/cps/rde/xbcr/SID-3D8CCB6A-2E5763F4/04/Telecom_PA-001004-00-03_XML_Development_Guide_Release_1.4.2.zip

--
Matt

On Mon, Mar 16, 2009 at 1:34 PM, Steve Davies davies...@gmail.com wrote:

 2009/3/16 David Ruggles da...@safedatausa.com:
  Is it possible to control the light on a programmable button without the
 blf
  option? I'm using a programmable button to turn call recording on and off
  and I would like the light to indicate the status.
 
  Thanks,
 

 9133i phones are pretty much obsolete, and are not getting firmware
 updates, so I do not know whether Aastra ever put any of their XML
 application control code into that model. If they did, then it should
 be possible to respond with button status using XML updates from the
 server, otherwise you'd need to upgrade to one of their currently
 supported phones, which are almost certainly capable of this sort of
 thing.

 PS. I have never personally used the XML facility of Aastra phones,
 but I hear quite good things about it.

 Regards,
 Steve

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Re: [asterisk-users] building asterisk-1.6.0.6 failed!

2009-02-23 Thread Matt Watson
I find it a little strange that for some reason your box is using includes
located in /usr/local... while there could be reason for this, that seems
like a sign that something might be a little broken on your box.

Also, if you don;t mind me asking...

why would you want to install * directly in /usr?  I could undersatnd if you
are building a distribution package or something, but personaly, i would
install to /usr/local or even some special place just for * just to help
keep the box more organized.

--
Matt

On Mon, Feb 23, 2009 at 7:17 PM, Tamer Higazi th9...@googlemail.com wrote:

 Hi!
 I have problems building asterisk 1.6.0.6.

 ./configure --prefix=/usr
 make

 gets me:

 enerating embedded module rules ...
   [CC] extconf.c - extconf.o
 In file included from /usr/local/include/datatypes.h:50,
 from /usr/local/include/err.h:49,
 from extconf.c:45:
 /usr/local/include/integers.h:50:67: error: srtp_config.h: No such file
 or directory
 In file included from /usr/local/include/datatypes.h:50,
 from /usr/local/include/err.h:49,
 from extconf.c:45:
 /usr/local/include/integers.h:103: error: conflicting types for 'uint64_t'
 /usr/include/stdint.h:56: error: previous declaration of 'uint64_t' was
 here
 make[1]: *** [extconf.o] Error 1
 make: *** [utils] Error 2


 Now, I think this is only a dependency problem. could anyone of you tell
 me, which and where I am able to get the missing sources to successfully
 compile asterisk?!


 Tamer

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Re: [asterisk-users] HDD FULLL

2009-02-23 Thread Matt Watson
On Mon, Feb 23, 2009 at 9:38 PM, Steve Edwards asterisk@sedwards.comwrote:

 On Tue, 24 Feb 2009, David @ULC wrote:

  When I am trying to delete voice logs,
  [r...@vicidialnow monitor]# rm * -r -f
  -bash: /bin/rm: Argument list too long

 In the past 30 days, you've asked questions about

 configuring Apache to process PHP files,
 Vicidial,
 Ntework Cards,
 Auto Detecting hardware,
 BAT files on CentOS,
 Root Password not taking,
 How to find a file,
 Looking for a Free VOIP Billing and Soft Switch,
 What is a VPN,
 How do delete files,
 oh, and a couple of Asterisk questions.

 This is an Asterisk users list. We're here to help each other with
 Asterisk questions and problems, not to be your personal, for free, life
 coach.

 If you are being paid to work on an Asterisk system, you are in over your
 head. You are defrauding your boss and most likely will give him and
 everyone in the company a bad impression of Asterisk.

 Continuing to answer your questions will only continue to enable you.

 Please take a step back, buy some books, take some courses, practice on
 your own systems on your own time.



Just have to say it: I totally agree with you and was thinking the same
thing as soon as I read that they tried rebooting the machine thinking that
would free up space on the HD... didn;t even realize this was the same
person that had posted some of those other threads

--
Matt
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Re: [asterisk-users] zaptel telephone cards and asterisk in another pc

2009-02-20 Thread Matt Watson
Your boss is going to change their mind when they see how awful and
unreliable this setup is going to be lol.



On Fri, Feb 20, 2009 at 1:42 PM, Ignacio sanfermi...@gmail.com wrote:

 yep, it is mainly due to cabling issues. My boss doesn't want to
 recabling the office.


 On Fri, Feb 20, 2009 at 7:19 PM, Jeff LaCoursiere j...@jeff.net wrote:
 
  Why do you want the cards in the client machines instead of the main
  asterisk server?  Cabling issues?  This sounds like it will be a horrible
  spaghetti mess...
 
  j
 
  On Fri, 20 Feb 2009, Ignacio wrote:
 
  Thank you very much for your fast answer Eric.
 
  I was trying to avoid to have to install as many asterisk as pcs I
  have. But I think there is no way to do it. I only have seen something
  like network block device, but not sure if it is going to work and
  quite difficult to configure properly.
 
  Anyway I think the fast and easier way will be installing and asterisk
  in every client.
 
  Thanks again.
 
  On Fri, Feb 20, 2009 at 6:41 PM, Eric Wieling, Asteria Solutions Group
  ewiel...@asteriasgi.com wrote:
  Ignacio wrote:
  I have some zaptel cards, and I would like to install them in some
  user's computers. Is there any way to connect those cards with
  asterisk server (which is in another computer)?
 
  All manuals I have read explain how to connect asterisk and zaptel
  cards in the same computers, but not on different ones.
 
  You can install Asterisk on the PCs with the cards in them and then use
  SIP or IAX2 to transport those calls to your main Asterisk server.
 
  --
  Eric Wieling * Asteria Solutions Group * Huntsville, AL
  Call centers * IVRs * Enterprise PBXs * Conferencing applications
  256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com
 
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Re: [asterisk-users] TAPI and Asterisk

2009-01-30 Thread Matt Watson
There is a Digium solution to this as well...

Digium recently acquired snap-a-number and has put own their own version of
it already called Asterisk Desktop Assistant

It does exactly what you are looking for plus some more... Information is
still a little scarce since it is pretty new, but there is some limited
information as well as some manuals that should give you enough information
to get it going (its actually very easy).


http://forums.digium.com/index.php?c=8sid=0a71abedc29238acee530a0461c914b0

Oh yeah, its also free (as in beer).  Though I am not clear if its going to
stay that way or not... But I haven;t seen anything that would indicate
otherwise.


--
Matt Watson


On 30/01/09 11:11 AM, Jeff LaCoursiere j...@jeff.net wrote:

 
 Following up my own thread, I am kicking myself for quickly posting
 without doing a bit of research.  Apparently (no surprise) this
 integration of Outlook and Asterisk is very old news, and there are many
 projects out there.  Anyone dealt with Thirdlane?
 
 http://www.thirdlane.com/products/thirdlane-dialer
 
 This seems to be just the ticket...
 
 Cheers,
 
 j
 
 On Fri, 30 Jan 2009, Jeff LaCoursiere wrote:
 
 
 Funny how a topic will come up that you have never dealt with before, and
 suddenly it comes up from multiple directions at the same time.  I was
 recently involved in a meeting where TAPI (which I understand only
 vaguely) was proposed as way to link a custom application to Asterisk for
 outbound and inbound call processing, much like SugarCRM and probably
 others are doing.
 
 Today I was asked by an existing client if I knew a way to synch their
 mobile device contacts with the system in some way so that they would have
 quick access to speed dial or otherwise call up a personal directory on
 their (Polycom) phones that could be synched in this manner.
 
 It struck me that the Polycom directory interface is a bit kludgy for such
 things, having no search capability and no sorting capability once loaded
 that I am aware of.  Given the meeting last week it seems that a more
 elegant solution would be to link Outlook itself with Asterisk, so right
 clicking a contact makes it possible to launch an outbound call.  That
 would take care of integrating a WHOLE LOT of devices, as (sadly) the MS
 contact database would be the go-between that all of these devices synch
 with in one way or another already.
 
 Is TAPI the right protocol to investigate for this purpose?  Would
 something like Fonality's HUD software bridge this gap?  Has this
 wheel already been invented?
 
 Hoping for some thoughts!
 
 Cheers,
 
 j
 
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Re: [asterisk-users] Asterisk 1.6 T38 to G711 transcoding is this possible?

2009-01-23 Thread Matt Watson
On Sat, Jan 17, 2009 at 11:51 AM, Steve Gladden 
aster...@michiganbroadband.com wrote:

 The scenario we have is fax send/recieve software that ONLY talks T38
 and an asterisk box.

 We have ITSP providers that do NOT talk T38 but G711 only.

 Does asterisk have the capability to take the T38 call from an ATA
 or T38 software then bridge/transcode it and do G711 out to the PSTN
 providers?

 If not is there another product PAID or FREE software or hardware that can
 do this easily and reliably?

Even if you find something that claims to be capable of doing this, it will
not work reliably.  T.38 is a protocol designed specifically to overcome the
challenges of faxing over IP networks.  Without it, depending on how good of
a network and route you have to your ITSP, your results will either be:

A) won;t work at all due to unpredictable network connections
B) Will work sometimes occasionally most of the time - all of which
are not suitable answers in an business environment. It will almost
certainly never be works flawlessly

In short, you cannot fax reliably over IP without T.38... you might want to
get an additional ITSP account with another provider and use it strictly for
faxes.  I;m sure there are lots that support T.38, but the only one I know
that a friend uses is Gafachi.  Your other option is to get a POTS line into
your * box, in which case (I believe) Asterisk 1.6 has the ability to act as
a T38 Gateway for this purpose.  It does not exist without 3rd party modules
for 1.4 I believe though... unless somebody has backported all the new T.38
stuff in 1.6.

--
Matt
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Re: [asterisk-users] Root Password not taking

2009-01-22 Thread Matt Watson
On Thu, Jan 22, 2009 at 11:32 AM, Max Brooks m...@legatio.com wrote:

 Jim Dickenson wrote:
  What I have done in the past to set the password for root is to boot
  in rescue mode and edit /etc/shadow setting the password to some know
  value from another system.
  --
  Jim Dickenson
  mailto:dicken...@cfmc.com
 
  CfMC
  http://www.cfmc.com/
 
 I personally prefer to chroot into the / partition and run passwd.


Yep, thats pretty much the best way, and more or less one of the only
methods that is going to work regardless of Linux distribution, or other
UNIX variant.

Many distros now, like most of the UNIXs actually still require your root
password when booting single user mode - as they should.  Gaining root
access to a system even with physical access to the machine *should* be more
difficult than simply picking a different grub boot option.  I realize that
is not the case across all distros, but IMO it should be.

For distros that do require a root password when booting single user mode,
your only real options have already been mentioned here...

1) boot from a CD, mount your partitions then:
   a) manually edit /etc/shadow (Linux only) and change
the field containing the encrypted password with another encrypted password
that you know what the uncrypted version is
   b) chroot into your mounted partitions and then run
passwd as normal (this should be work almost all UNIXs)

(b) is the more generic and preferred method IMO - it should work just about
everywhere... unless you have total disk encryption or encrypted filesystems
and are unable to mount the partitions... in which case... best of luck to
you.

--
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Re: [asterisk-users] random Linksys question

2009-01-22 Thread Matt Watson
Yes, it is available on the SPA2102 - you just login to the web interface,
goto the advanced section, then lan setup... its the very first option.

--
Matt

On Thu, Jan 22, 2009 at 3:29 PM, Kristian Kielhofner 
kristian.kielhof...@gmail.com wrote:

 On Thu, Jan 22, 2009 at 3:11 PM, Jeff LaCoursiere j...@jeff.net wrote:
 
  Can you configure the LAN port on the back of a 2102 to be bridged
  rather than routed to the WAN port?
 

 To my knowledge this is available on all Linksys ATA type devices that
 offer both ports.

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

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Re: [asterisk-users] Root Password not taking

2009-01-22 Thread Matt Watson
On Thu, Jan 22, 2009 at 1:48 PM, Wilton Helm wh...@compuserve.com wrote:

  There have been a number of answers provided.  The one that was given to
 me when I encountered this same problem was to boot a live CD, mount the
 root file system and delete the password file which would force your normal
 distro boot to request a new root password next time.


I'd highly recommend to NOT simply delete your password files (/etc/passwd 
/etc/shadow)... you are deleting a lot more than just your compromised your
root password by deleting these files.  You are also deleting every other
account that is on your system (ie. you should be running Asterisk under its
own account and not as root!).  Of course it depends on what other things
are running on your server, but you cannot reasonably expect your distro to
keep track of every change you have ever done to these and re-create it.  It
might give you a skeleton version like you are suggesting, but you are
certainly going to lose data.


Step 3 was replace a few key executables like ps so I couldn't do
 administrative tasks


I can't really speak about your case specifically, but generally replacing
certain executables like ps, netstat, login, w, who, etc. etc. are not done
to prevent administrative tasks - infact, a person that actually knows what
they are doing with a rootkit doesn;t actually want to prevent you from
doing administrative tasks.  Generally they don't want you to know at all
that they have broken into your system.  The reason why these binaries get
replaced is to try and cover their tracks.  for example, they replace 'ps'
so that it will not display certain processes running on the system (making
it harder for you to even notice), the same with netstat, it will not
display network connections to certain IPs, 'login' gets replaced with a
version that has a hard-coded login  password that will grant root access
to your system without knowing (or caring) what the real root password is,
and might also not do a few things that are part of the normal login
process... like writing to UTMP / WTMP so there is no record of the login.


The problem is that you don't know exactly what files have been changed and
 if they have left a trap door or something.  You could fix the root
 password, and even discover and restore a few key files, only to find it
 hacked 5 minutes later because you didn't know everything that had been
 altered.  For that reason, few people will put a system back on line after
 the root password has been compromised.  Re-installation is the only safe
 way.  If some of your directories like /home and /user have separate mount
 points, they don't have to get wiped out in the process.


This is getting a little off topic... but there actually are ways of
determining exactly which files have been changed.  Stuff like 'tripwire'
has been around for like a decade and is designed to do exactly this...
everytime you run it, it will tell you every single file that has changed.
It does this by creating hash of every file when you run it and then
compares those hashs to previously stored values.  If the hashes do not
match, then something has changed and gets reported.  Of course hashes can
be defeated, it is however not entirely easy, and way beyond what your
typical script kiddie hacker is going to be capable of doing.  And of
course, that is just what trip wire does... there are plenty of other
intrustion detection systems that I;m sure do things a little differently.
Some other things like FreeBSD used to (maybe still does?) come with some
basic security stuff that was installed by default and emailed you daily
reports about things that changed.  Like any new binaries it found on the
system that were setuid root (ie. cp /bin/bash /var/spool/.myhiddenrootshell
; chown root /var/spool/.myhiddenrootshell ; chmod u+s
/var/spool/.myhiddenrootshell).  Something like FreeBSD's scripts were
intelligent enough to find basic things like that.

That being said, I do agree with you, re-installation in 99% of cases is a
superiour option when compared to trying to restore a compromised system.
Typically its much quicker too.
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Re: [asterisk-users] Root Password not taking

2009-01-22 Thread Matt Watson
On Thu, Jan 22, 2009 at 1:52 PM, David @ULC ucoms2...@gmail.com wrote:


 I tried :

 1. Shut down the machine. (Ctrl+Alt+Del)

 2. When it reboot and reach the CentOS boot up screen, then press any key
 to go into a select menu. Then press e and navigate to the second line
 grub.conf line (kernel) and press e to edit the line to

 option 1 : SPACE 1 ( 1″) at the end of the line. OR
 option 2: SPACE S ( S) a the end of the line OR
 option 3: add single to ro root=LABEL=/ single



Your error is at option #3, the root= line should be pointing to the
partition of your root (/) filesystem.  You are supposed to replace LABEL
with it - typically this is going to be something like /dev/sda1  (if you
are SCSI or SATA - /dev/hda1 if its IDE)... though I believe CentOS uses
initrd, and it might be a little different when using initrd, i;m not very
familiar with it so hopefully somebody else will chime if it should be
something different.  but it also may not be sda1... it could be sda2, sda3,
sdb1, sdb2, sdz4, hdg3... in otherwords, too many options to list, this is
specific to your system - but sda1 would be typical.





 Then hit ENTER and press 'b' to reboot.

 After it reboot, and stop at '# command line, type passwd to create the
 new root password.

 Reboot the machine as usual and access your root with new password.


 On Thu, Jan 22, 2009 at 9:22 PM, David @ULC ucoms2...@gmail.com wrote:


 In one of my center , its not taking root password.

 Anyways to recover it ?

 In other terms , I lost the control of server.

 Any solution or re-installation is the only way left ?

  I am using CentOS.



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Re: [asterisk-users] Root Password not taking

2009-01-22 Thread Matt Watson
On Thu, Jan 22, 2009 at 5:30 PM, Wilton Helm wh...@compuserve.com wrote:


 Tripwire would be fine, if it had a baseline, but I don't think its any
 good after the fact.


Correct - tripwire does need to be setup beforehand, and its not the most
simple thing to setup *properly*.  After the fact... you are basically out
of luck unless you are using a binary-based distribution and want to
re-download all the packages and compare the hashs... but thats simply too
time consuming and reinstalling is faster and more reliable.

--
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Re: [asterisk-users] Root Password not taking

2009-01-22 Thread Matt Watson
On Thu, Jan 22, 2009 at 6:05 PM, Wilton Helm wh...@compuserve.com wrote:

  making sure to patch any holes through which the hacker might have come

 In my case, I had been getting regular attacks through SSH for months,
 probably 100 a day (bots).  Apparently after nine months of this, someone
 stumbled on to my password which regrettably was composed of two dictionary
 words with no special characters, making it susceptible to dictionary
 search.

 When I re-installed, I put SSH on a non-standard port and haven't had a
 single attempt to attack the system since.



While its certainly not a good idea to rely on... security through obscurity
can work very well :)


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Re: [asterisk-users] DAHDI install dont need download of echo cancel

2008-12-18 Thread Matt Watson
after you have configured zaptel manually the first time, copy the
menuselect.makeopts file that is generated in the root directory of the
zaptel source to a file /etc/zaptel.makeopts.

presumably this is available for people that have moved on to DAHDI as well,
and I would guess it should be /etc/dahdi.makeopts - but I have not verified
that.

--
Matt Watson


On Thu, Dec 18, 2008 at 11:49 AM, Jerry Geis ge...@pagestation.com wrote:

 
  Jerry Geis schrieb:
  / Is there a way to install DAHDI and NOT download the echo canceler
 files?
  // I dont have firewall access and its failing.
  // I dont need the files as there is no card installed.
  //
  // How do I get past this?
  /
  If I remember correctly you can un-check them in
  `make menuselect`.
 
 
 Philipp Kempgen
 
 Thanks I did not look there.

 Is there a way to do it in an automated fassion. (doing more in the
 future).
 Dont want to have to manually change menuselect.

 Jerry

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Re: [asterisk-users] dahdi-monitor in France

2008-12-11 Thread Matt Watson
1. Ask your telco, they probably have them, but you may have some difficulty
in finding somebody at your telco that has a clue about what you are talking
about.  You can find some lists doing some google searches for the numbers
and hope to get lucky... but as far as I know, there is no official
repository for these test numbers.

2. I wouldn;t use an overseas number personally... those calls are certainly
getting encoded / decoded and reencoded several times, and more than likely
getting compressed, all of which is going to have an impact... it *might* be
better than nothing... but i would expect very poor results.

3. You are right, you can';t really just make one yourself from scratch, you
need a source that has already been tuned properly to use as a reference for
creating your own.


--
Matt Watson


On Thu, Dec 11, 2008 at 11:01 AM, Olivier [EMAIL PROTECTED] wrote:

 Hi,

 I would like to tune rx/tx gains using dahdi-monitor for a system which
 will be connected to french PSTN.
 I'm not aware of any public phone number in France I could call to get a
 normalized 1004Hz signal.


 My questions are :
 1. Does such numbers exist ? Is there a directory somewhere listing some of
 them ? Do you think regulations could make providing such numbers mandatory
 for (some) Telcos ?
 2. Does it make to use a number aboard instead if I can't find any local
 ones ? I don't think so, but I prefer to check.
 3. I can't imagine a process allowing me to create my own (chicken and egg
 problem). Is it correct ?

 Regards

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Re: [asterisk-users] Problem with Aastra 480ci and qualify=yes

2008-08-18 Thread Matt Watson
I;m using Aastra 480i's 9133i's, 9112i's, and 57i's and none of them have
experienced problems with qualify=yes.

I;m currently on Asterisk 1.4.17, but I've also tested them with 1.4.14 up
to 1.4.19.

--
Matt
http://www.mattgwatson.ca

On Fri, Aug 15, 2008 at 10:59 AM, Drew Gibson [EMAIL PROTECTED] wrote:

 James Lamanna wrote:
  Hi,
  We have a few Aastra 480ci phones and we've noticed that in order to
  get the phone to receive a call, qualify must be = no.
  Apparently the Aastras do not respond to the qualify message (or
  respond in a way Asterisk doesn't understand) and Asterisk thinks the
  phone is unreachable.
  However, this now prevents MWI from working properly on the phones.
 
  Does anyone know how to get MWI working without qualify? Or how to get
  qualify working again with the Aastras?
 
 

 We have a number of 480i and one 480ct all setup with qualify=yes
 (Asterisk 1.2.24)
 Our inbound call centre seems to be pretty busy and my own MWI lights up
 far too often. Never had a problem with either.

 Which version of firmware?
 Which version of Asterisk?
 What's in your sip.conf?
 What error messages show on the console?
 Anything relevant in the logs?

 regards,

 Drew

 --
 Drew Gibson

 Systems Administrator
 OANDA Corporation
 www.oanda.com


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Re: [asterisk-users] Visual Dial Plan

2008-07-27 Thread Matt Watson
I've seen it before infact there is a website setup where people can
post stuff made with it... kind of super nerdy!
http://www.ratemydialplan.com

--
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On 7/27/08, Peter Lindquist [EMAIL PROTECTED] wrote:

  Dean Collins wrote:

  I just stumbled across this on youtube.



 Does any on the list us it? This is the first I've heard over it.



 http://www.youtube.com/watch?v=H1j5OrgL1og





 Regards,

 Dean Collins
 [EMAIL PROTECTED]

 +1-212-203-4357 (New York)
 +61-2-9016-5642 (Sydney)
 http://www.Cognation.net http://www.cognation.net/profile



 Yes, I use it and it is a great tool I think. If anything I do miss an
 ability to print out the graphical representation of the dial plans in the
 current version - this is being worked on though.

 Best regards,

 Peter Lindqvist
 Voxion Ltd.


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Re: [asterisk-users] Not a valid SIP contact - Asterisk 1.4.21.1 Mitel SIP phones

2008-07-19 Thread Matt Watson
On July 19, 2008 11:22:08 am Mark Wiater wrote:
 Hi,

 I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1
 Asterisk server (and a couple of previous 1.4 versions). They're
 mostly happy with the combination except for this one issue.

 For incoming calls only, either originating from other local SIP
 phones or from a PRI, calls won't get bridged (remote party get's
 hung up) if the call is answer too quickly on the Mitel. Or so it
 seems. The receiving Mitel phone thinks the call is in session though.

 Asterisk is reporting errors like:

 [Jul 19 10:46:41] NOTICE[2466]: chan_sip.c:8068
 set_address_from_contact: '72.16.1.20;tag=as7b9f4bfb' is not a
 valid SIP contact (missing sip:) trying to use anyway
 [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:8097
 set_address_from_contact: Invalid host name in Contact: (can't
 resolve in DNS) : '72.16.1.20'
 [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination:
 Can't find address for host '72.16.1.20'
 [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination:
 Can't find address for host '72.16.1.20'


Might want to post a sip debug of one of the sessions from the Mitel phone.


-- 
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http://www.mattgwatson.ca

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Re: [asterisk-users] Digium PRI and Echo cancellation

2008-07-17 Thread Matt Watson
On July 16, 2008 08:01:38 am Loic Didelot wrote:
 Hello,
 I would like to double check what Echo Cancellation my Digium Card uses.

 I thought I bought the little more expensive card that integrates
 EchoCancellation. How can I check?

If you load the modules with debug=1  (maybe this appears without it too... 
not sure):

Apr 23 12:52:55 nelson VPM400: Not Present
Apr 23 12:52:55 nelson VPM450: echo cancellation for 64 channels
Apr 23 12:52:55 nelson VPM450: hardware DTMF disabled.
Apr 23 12:52:55 nelson VPM450: Present and operational servicing 2 span(s)
Apr 23 12:52:55 nelson Completed startup!

Thats on my TE220B.  the VPN450 is the hw echo can daughter board.


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Re: [asterisk-users] Digium PRI and Echo cancellation

2008-07-17 Thread Matt Watson
I believe HPEC actually is the same algorithm (G.168) that the HW echo
cancel modules use.. the difference being that HPEC uses up CPU cycles and
its performance will be impacted on a system with higher CPU load, whereas
the HW modules have a dedicated DSP for it.

http://blogs.digium.com/2007/09/06/fun-with-hpec/


--
Matt

On Thu, Jul 17, 2008 at 10:00 AM, Noah Miller [EMAIL PROTECTED]
wrote:

 Hi Loic -

  According to that its using MG2.

 I think it will say MG2 regardless of whether or not there is a
 hardware module present.


  Shouldnt it be using something like
  HPEC?

 I don't think the hardware echo cancellers use the HPEC algorithm.  As
 Eric and Matt have mentioned, dmseg will tell you if a hardware echo
 cancel module is being loaded.


 - Noah

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Re: [asterisk-users] Incoming calls on zaptel not answered.

2008-07-15 Thread Matt Watson
On July 14, 2008 08:24:33 pm Jose Flores Galicia wrote:
 After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri,
 zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop
 working.

THis isn;t going to fix your problem... but just FYI, you don't need to 
install libpri if you are just using a TDM400P (since its not a PRI / BRI 
[1.6 libpri does BRI as well] card). 

Might save you a little bit of time in the future, and its one less thing to 
consider as a problem.

-- 
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Re: [asterisk-users] (announce) asterisk T.38 gateway

2008-07-10 Thread Matt Watson
I'd probably be a little pissed if I were Steve Underwood if somebody
pocketed over 10k $USD for taking credit for a product that my free library
did the bulk of the work for.

I don;t think i'd feel that the entire bounty should be mine - after all
there would of been nothing stopping me from doing it myself... but credit
should be given where credit is due.   Even if its just something as a sign
of appreciation.

Given that spandsp is GPL'd Steve obviously never intended to make a ton of
money off of it... but I;m sure he'd love to receive something for his work,
or use some of that money to further develop spandsp.

That being said... i;m also quite pleased to see T.38 support being worked
on for Asterisk... its a pretty important area to further develop IMHO.

--
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On Thu, Jul 10, 2008 at 11:54 AM, Steve Totaro 
[EMAIL PROTECTED] wrote:



 On Thu, Jul 10, 2008 at 11:43 AM, Steve Totaro 
 [EMAIL PROTECTED] wrote:



 On Thu, Jul 10, 2008 at 10:24 AM, Steve Underwood [EMAIL PROTECTED]
 wrote:

 Vinícius Fontes wrote:
  When people release software under the GPL license, like Steve
 Underwood did with libunicall, spandsp and so on, they were supposed to know
 that other people has the right to use their code.
 
 The problem is that almost any licence term which tries to limit the
 obnoxious behaviour of other people has too many unpleasant side
 effects. GPL 2.0 is the best compromise I've found, so that is what I
 used for everything unless recently. To make my stuff licence compatible
 with FreeSwitch I recently relicenced most of my work as LGPL 2.1. This
 is having undesirable consequences, though. Its really a tough issue,
 and GPL 2.0 showed immense foresight in just accepting the non-existence
 of perfect solutions. GPL 3 seems to have forgotten the lesson somewhat.

 Most of the time I just want to give up producing anything at all.

 Steve


 So are you angry that he may gain monetarily from your your work, or is it
 hurt pride that he is basically taking credit for it?

 The answer to that should guide you in how you release your work in the
 future.

 Thanks,
 Steve Totaro


 I also want to add that if someone asked me to name the top five names that
 came to mind when thinking of Asterisk, Jim Dixon, Mark Spencer, Steve
 Underwood, Nicolas Gudino, and I will leave off the fifth as to not leave
 anybody out ;)

 Thanks,
 Steve

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Re: [asterisk-users] cell phone hangup not getting recognised by system

2008-07-09 Thread Matt Watson
is it only cell phone calls that don't work?  or is it any external call
coming in over your lines?

What type of inbound lines do you have?  I;m guessing analog lines... if
thats the case what type of signalling are you using?

if its only cell calls and not all external calls then I have no idea what
it'd be... but if its all calls then its probably a signalling problem...
you might be using loop start when you should be using kewlstart... or it
might be that you need to get disconnect supervision added to your lines
(essentially converting loop start - kewl start)

--
Matt
http://www.mattgwatson.ca


On Wed, Jul 9, 2008 at 3:08 AM, Giorgio Incantalupo 
[EMAIL PROTECTED] wrote:

 Hi Kate,
 have you tried the busydetect parameter in zapata.conf?
 Take a look here for other useful parameters:
 http://www.voip-info.org/wiki-Asterisk+config+zapata.conf

 Giorgio.


 Lists wrote:
  Hi all,
 
  When I do a test call into the box (which is running latest version of
  Trixbox) it all goes fine. If i decide to hangup the cellphone (during
  the ivr playing options) the system does not recognize the hangup and
  the system continues through and ends up at the timeout option.
 
  What settings do I need to change to fix this. Is it the rxgain? If so
  is there something i can use to figure out what the gains should be set
 at?
 
  Thanks
  Kate
 
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 [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu
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Re: [asterisk-users] New Polycom SpectraLink 8002 Wifi SIP Handset

2008-07-06 Thread Matt Watson
On July 5, 2008 09:00:57 pm Julian Yap wrote:

 Nice review.  Can this phone be provisioned without using TFTP?  For
 example, over the internet?  Are there other provisioning methods?

Theres no reason that you can;t do TFTP over the internet... Not so certain 
i'd recomend it since TFTP is authentication-less, but its still possible... 
and if security is in mind you could limit your TFTP server to specific 
source IPs

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Re: [asterisk-users] Cell phone to PSTN adapter or IAX

2008-07-05 Thread Matt Watson
On July 5, 2008 01:50:52 pm Joseph wrote:
 Are they any such things as Cell phone PSTN adapters?

 Openmoko is coming out so I hope it will be possible to register it to
 Asterisk and/or there will be a small iax registration program to
 communicate with asterisk.

The beauty of the OpenMoko is that it is entirely open and completely 
hackable... they encourage people to modify it and write software for it... I 
imagine making a SIP/IAX client for it would probably be rather easy.

That being said... since the OpenMoko runs Linux as its OS... its probably 
possible to run Asterisk on the OpenMoko... you could probably even use 
Asterisk as that SIP/IAX client using chan_alsa to access the built in 
speaker/microphone... (I;m assuming they are accessible through ALSA).

I don;t know the OpenMoko inside and out... i;ve done some light reading on it 
over the last year or so... so i;m really just making some logical 
assumptions here.

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Re: [asterisk-users] Centos-5.2 and zaptel-1.4.11 do not get along well

2008-06-30 Thread Matt Watson
On June 30, 2008 06:25:17 pm Mark G. Thomas wrote:
 Hi,

 After doing a yum update on my previously Centos-5.1 system, now
 zaptel-1.4.11 fails to build with this below.



   CC [M]  /opt/src/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o
 In file included from /opt/src/asterisk/zaptel-1.4.11/kernel/xpp/xpd.h:26,
  from
 /opt/src/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.c:27:
 /opt/src/asterisk/zaptel-1.4.11/kernel/xpp/xdefs.h:117: error: conflicting
 types for ‘bool’
 include/linux/types.h:36: error: previous declaration of ‘bool’ was here

I;m not sure but I think somebody about a month ago had a very similiar 
compilation problem... infact I think it was the same error.

I;m not sure what the proper way is to fix your CentOS box is, however an 
option for you might be to just not compile the xpp module.  xpp I believe is 
the Xorcom AstriBank... if you don;t actually have an AstriBank then there is 
no sense in even compiling/installing the drivers for it.

I;m guessing you haven't run a make menuselect to select only the drivers you 
need?

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Re: [asterisk-users] Disto choice for Asterisk with AVM Fritz!PCI cards

2008-06-30 Thread Matt Watson
On June 30, 2008 08:44:44 pm Simon wrote:
 Hi There,

 I am looking to build an Asterisk server with dual AVM Fritz!PCI cards
 linked to 2 BRI in New Zealand. Just wondering if anyone has done
 this, and if you have any ideas about the best disto choice for this
 task?


Let me be the first to say: Pick whatever distro you are comfortable with

Distro is more of a personal choice than anything... ultimatly they all have 
the same software available to them (for the most part), they all just do it 
a little bit differently.

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Re: [asterisk-users] No Codecs and app

2008-06-24 Thread Matt Watson
On June 24, 2008 01:57:45 am troxlinux wrote:
 Hi list, recently install asterisk 1.4.21 in a centos 5, and after
 having installer the zaptel 1.4.10.1 and libpri 1.4.4 I don't see in
 the directory  module any codec, and neither  app.

 almost install all the asterisk options

 this worries to me !

 alone I see these packages inside the directory



 app_addon_sql_mysql.so  cdr_addon_mysql.so   res_config_mysql.so
 app_saycountpl.so   chan_ooh323.so  format_mp3.so

 some help that they can provide me?


Looks like you have installed asterisk-addons and not asterisk itself... if 
you compiled from source maybe you just forgot to 'make install' for 
asterisk?


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Re: [asterisk-users] Controlling cell phone VM / Fax waiting notification icon for asterisk VM

2008-06-23 Thread Matt Watson
On June 23, 2008 08:08:53 am OCG Technical Support wrote:
 I little more digging and I confirmed that cell phone VM and FAX waiting
 icons are in fact controlled by a proprietary SMS message format.  Here's
 what I found:





 Message Waiting Indication Group: Store Message
 This Group allows an indication to be provided to the user about the status
 of types of
 message waiting on systems connected to the GSM PLMN. The mobile may
 present this
 indication as an icon on the screen, or other MMI indication. The mobile
 may take note of
 the Origination Address for messages in this group and group 1100. For each
 indication
 supported, the mobile may provide storage for the Origination Address which
 is to control
 the mobile indicator.
 Text included in the user data is coded in the Default Alphabet.
 Where a message is received with bits 7..4 set to 1101, the mobile shall
 store the text of
 the SMS message in addition to setting the indication.
 Bits 3 indicates Indication Sense:
 Bit 3
 0 Set Indication Inactive
 1 Set Indication Active
 Bit 2 is reserved, and set to 0
 Bit 1 Bit 0 Indication Type:
 0 0 Voicemail Message Waiting
 0 1 Fax Message Waiting
 1 0 Electronic Mail Message Waiting
 1 1 Other Message Waiting*
 * Mobile manufacturers may implement the Other Message Waiting indication
 as an
 additional indication without specifying the meaning. The meaning of this
 indication is
 intended to be standardized in the future, so Operators should not make use
 of this
 indication until the standard for this indication is finalized.





 Now the tough part...does anyone want to create an app to send notification
 to a cell phone to set/clear these bits?

could you provide a link to where you got the info from?  I'd be interested in 
seeing if i can get this to do anything useful.

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Re: [asterisk-users] Send cell phone #VM waiting, just like cell carrier

2008-06-22 Thread Matt Watson
On June 22, 2008 11:32:56 am OCG Technical Support wrote:
 Well, I realize that there must be some proprietary protocol between the
 carrier and the phone, since they have a dedicate spot on the cell screen
 for # VM waiting...

 As for an SMS message, is there a module/app which allows easy SMS
 messaging?  (I looked a couple of years ago but only found commercial
 modules)

Don't quote me on this, but I think that the standard voicemail notifications 
that cell phones recieve are just a regular SMS message but in some kind of 
special format that the phone recognizes and does not display as a regular 
text message.

I had an unlocked Moto razr at one point that didn't come from my cell 
provider, at one point while I had it the VM notifications didn;t really 
work... everytime i got a VM I'd get a SMS message with special characters in 
the message body.  I sort of chalked it up to not using the branded firmware 
that my provider would have put on my phone if the phone had come from them.

At any rate, if I'm right, and you can find some information on that protocol, 
it might turn out to be alot easier than you think.

and btw - Asterisk ships with app_sms but depending on your country it may not 
really be what you are looking for.  I could be very very wrong here, but I 
don't believe Asterisk has a great implementation for text messaging in North 
America.  This is probably because every cell carrier seems to do it just a 
little bit differently.

Depending on your needs however, most cell carriers offer email-to-SMS 
gateways so you can just send an email to 
yourcellnumber@yourcellcarrier.com kind of thing... if you need it to 
work from a machine with no Internet access however, then you might need to 
do something like use a dialup modem to dial into your cell carriers TAP 
number (provided they offer one and you can either explain to the tier 1 
support guy what you are looking for, or you can find the number on the 
internet).

However again, I don;t believe Asterisk has any interface for TAP, but in 
theory you could use something like the sendpage daemon to handle the TAP 
stuff and then have Asterisk hand off the text messages to sendpage using a 
simple bash script and the System() application... if Asterisk had an 
implementation for SNPP then you could send your messages to sendpage that 
way - also *some* cell providers even offer SNPP gateways over the internet, 
but again good luck explaining to the tech what you want.

I've actually been toying with the idea of creating a res_snpp or something 
for this purpose since I already use sendpage for our server 
monitoring/alerting system to page me on my cell via dialup TAP (so my 
monitoring server can still page me in the event of an internet outage).

Anyhow, sorry that message got a bit lengthy pretty fast!

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Re: [asterisk-users] Please Advice on Best High traffi c fxo gateway/cards

2008-06-15 Thread Matt Watson
On June 15, 2008 10:53:35 am Brian J. Murrell wrote:
 On Sun, 2008-06-15 at 17:43 +0300, James Mutuku wrote:
  Please advice on  channel bank

 Dude.  There's the cool new website you should check out.  It's
 www.google.com.

 Seriously.  This list is not full of people waiting to do the simplest
 research at your request.  Spend a few minutes and do some self-help
 before coming here asking the simplest, most general questions.  You are
 more likely to get answers to interesting questions rather than
 mundane-google-would-have-told-you-all-you-need-to-know-in-5-minutes
 questions.

 b/

Heh, there seems to have been a few of these kinds of requests on the list 
lately eh... I think I saw one a couple days ago where somebody was asking 
about a dozen very basic questions that they were tasked with to find out of 
Asterisk was suitable for their organization... and it was seemingly like 
they were literally trying to offload their job to this list

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Re: [asterisk-users] *OT* DLI Ethernet Power Controlle r $289 (I paid $200 for a two port websw itch)

2008-06-15 Thread Matt Watson
On June 15, 2008 12:11:13 pm Mark Hamilton wrote:
 This sounds good. Except I'm a little confused. Is this a reboot bar which
 uses Ethernet to do the reboots? Like a reboot bar, except in a PoE
 lifestyle?


Its just a PDU (power distribution unit) that has a web-interface (accessed 
via Ethernet)... it looks like it has the added bonus of having some RS232C 
ports that you can either attach a modem (to dial-into the device) or to 
connect to serial-console based equipment, like certain routers and switches 
so that you can access their serial console remotely.

Essentially its the equivalent of the APC AP7902 - 
http://www.apc.com/resource/include/techspec_index.cfm?base_sku=AP7902 it has 
a couple feature differences, but for the most part they do the same thing.

However, the cost is significantly less than than the APC model.  

I don't have any experience with either however.

All in all it looks like a decent product... i'd be interested in hearing from 
anybody that might of been using them for a long period of time (1-2yrs+).  
I'm pretty picky about power distribution, i've seen bad power cause too many 
problems in my computing history.

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Re: [asterisk-users] *OT* DLI Ethernet Power Controller $289 (I paid$200 for a two port webswitch)

2008-06-15 Thread Matt Watson
On June 15, 2008 12:10:03 pm Dean Collins wrote:


 Do you know if this unit has any power metering capability? I'd really
 like to start measuring which of my servers are using the most power etc
 and not sure from this description if this is possible.


Just FYI - the APC model I mentioned in my last post will do load metering... 
it'll cost you about twice as much as the one Steve posted however. The sale 
price that is, couple hundred more than the regular price.


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Re: [asterisk-users] *OT* DLI Ethernet Power Controller $289 (I paid $200 for a two port webswitch)

2008-06-15 Thread Matt Watson
On June 15, 2008 12:45:07 pm Steve Totaro wrote:
 If a PDU is just a power strip then this has many more bells and
 whistles.  My usage is being able to control power to those outlets to
 reboot or turn on/off equipment if it is hung or whatever other
 reason.

 Thanks,
 Steve T


Yep, a PDU is really just a broad term for anything that takes power input and 
distributes it to many devices - so yes a power strip/bar would be classified 
as a PDU...technically.  However I'd probably expect to get laughed at if I 
called a power bar a PDU!

There are pretty big differences between the $5 power bar you can buy at 
walmart and stuff that people typically put into server cabinets.

technically this device specifically would be a Switched PDU the switched 
part meaning it has the ability to turn on/off individual ports.  The 
interface to turning those ports on/off however is irrelavent to its device 
classification it could be a web interface like this device, a telnet/ssh 
interface, it might not even have any remote capability and just have 
physically switches for each port.

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Re: [asterisk-users] *OT* DLI Ethernet Power Controller $289 (I paid $200 for a two port webswitch)

2008-06-15 Thread Matt Watson
On June 15, 2008 01:05:40 pm Andres wrote:
 All in all it looks like a decent product... i'd be interested in hearing
  from anybody that might of been using them for a long period of time
  (1-2yrs+). I'm pretty picky about power distribution, i've seen bad power
  cause too many problems in my computing history.

 We have used the APC 8 port version for abour 5 years now.  It is rock
 solid.   The connected load metering is a very nice feature as well.
 The cost is about $380.

 Andres
 http://www.neuroredes.com



Thanks Andres,

I was actually was hoping to hear from somebody that has used the device that 
Steve orignally posted though!

I've used several other APC products and I don't think I'd need to think twice 
about buying anything else from them.

I do kind of think APC is a little like buying stuff from Sony... you pay a 
bit of a brand tax just to have the APC logo printed on it.  However I think 
that the APC logo on something means alot more than the Sony logo :)

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Re: [asterisk-users] Fax on FXS

2008-06-10 Thread Matt Watson
Ah, you got me there!  Could start throwing in a lot of Over's going down 
that road :)

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-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Tuesday, June 10, 2008 4:10 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fax on FXS

On Mon, Jun 09, 2008 at 06:53:34PM -0400, Matt Watson wrote:
 On June 9, 2008 01:34:31 pm Eric ManxPower Wieling wrote:
  You should not expect FaxOverVoiceOverIPOverInternet to work well.  If
  you stick to ulaw codec for the entire call, it might work well enough
  for your use, but it might not.

 Just as an FYI - you have too many Over's in your description

 FaxOverVoiceOverIP would make sense, but seeing as how IP is short
 for Internet Protocol, saying Internet Protocol Over Internet doesn;t
 make much sense...

Unless you use an openvpn  / ipsec tunnel :-)

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

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Re: [asterisk-users] Problems configuring a PRI...

2008-06-10 Thread Matt Watson

1. Ditch the channels= in zaptel.conf that doesnt belong there (you've done 
the channel config with the bchan= and dchan=
2. your span= should *probably* be 1,1 instead of 1,0   in zaptel.conf the 
2nd 1 indicates to use that span as a primary timing source
3. not that it should matter, but you don;t need the duplicate group=, 
signalling=, switchtype= in zapata.conf
4. you can ditch rxwink= that setting is for non-PRI T1s

try that and see if that helps... I suspect the span not being used as primary 
timing source is whats causing your greif.

good luck!

-- 
Matt Watson
http://www.mattgwatson.ca

On June 10, 2008 05:22:40 pm Christopher Hoff wrote:
 I'm trying to get a Qwest PRI configured and working with my lab
 Asterisk server. They said that the switchtype is 5ess and the signaling
 is pri_cpe. My entries into zaptel.conf are:

 span=1,0,0,esf,b8zs
 bchan=1-23
 dchan=24
 loadzone = us
 defaultzone=us
 channels=1-23


 And my entries in zapata.conf are:

 language=en
 context=telco-incoming
 switchtype=5ess
 signalling=pri_cpe
 rxwink=300
 usecallerid=yes
 hidecallerid=no
 callwaiting=no
 usecallingpres=yes
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 canpark=yes
 cancallforward=yes
 callreturn=yes
 echocancel=yes
 echocancelwhenbridged=yes
 rxgain=0.0
 txgain=0.0
 callgroup=1
 pickupgroup=1
 immediate=no
 group = 1
 switchtype = 5ess
 signalling = pri_cpe
 group = 1
 channel = 1-23

 I'm not able to make/receive calls, and the error I'm receiving is:

 [Jun 10 11:32:37] WARNING[31768]: chan_zap.c:2393 pri_find_dchan: No
 D-channels available! Using Primary channel 24 as D-channel anyway!
 == Primary D-Channel on span 1 down

 Qwest says that the PRI is fine. I have a green light on the PRI card.

 Help!



 ___



 Chris Hoff

 Telecommunications Administrator

 SEI LLC

 Voice  +1 701 298 8865 Ext 2189

 Mobile +1 701 361 5976

 Fax +1 701 298 8860

 Email [EMAIL PROTECTED]




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Re: [asterisk-users] Fax on FXS

2008-06-09 Thread Matt Watson
On June 9, 2008 12:57:11 pm John Morey wrote:
 I've been thinking about something around these lines that I'd like
 feedback on.  What I'd like to d,o if it works, is have a fax machine in
 St. Louis connected up to my asterisk box in Atlanta via Internet/SIP so
 that anytime the fax machine in St Louis sends a fax it actually goes out
 through the asterisk box in Atlanta.  Something if I understand it
 correctly like : Fax-SIP(long distance)-Asterisk-FXO-Customer Fax. 
 Would something like this work?


This will not work:

Fax - SIP ATA - [internet] -  Asterisk - FXO - Fax 

Because you don;t have end-to-end T.38 support, Asterisk supports T.38 
pass-thru but not origination/termination (yet).

However, what *should* work is:

Fax - SIP ATA - [internet] - Asterisk - SIP ATA - Fax

the SIP ATAs obviously have to have T.38 support - for example a Linksys 
SPA2102 should do it for you.  I've never tried faxing between ATA's so I 
don;t know if they can actually negotiate T.38 support between each other, 
but I don't really see a reason why they couldn't.


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Re: [asterisk-users] Fax on FXS

2008-06-09 Thread Matt Watson
On June 9, 2008 01:34:31 pm Eric ManxPower Wieling wrote:
 You should not expect FaxOverVoiceOverIPOverInternet to work well.  If
 you stick to ulaw codec for the entire call, it might work well enough
 for your use, but it might not.

Just as an FYI - you have too many Over's in your description 

FaxOverVoiceOverIP would make sense, but seeing as how IP is short 
for Internet Protocol, saying Internet Protocol Over Internet doesn;t 
make much sense...

I would interpret that description as somebody trying to send a fax through a  
VoIP system using a voice-codec like G.711

However, there is the T.38 protocol which is designed to solve this exact 
problem, Asterisk support for it is just rather limited currently (pass-thru 
only).  

T.38 often gets referred to as FoIP (Fax over Ineternet Protocol)

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Re: [asterisk-users] Interoffice phone setup

2008-06-09 Thread Matt Watson
On June 9, 2008 07:49:13 pm Joseph L. Casale wrote:
 What type of PBX hardware do you have on-site? Also what make/models of
 phones?

 Michael/Darryl,
 I do have a local asterisk box, which is why I am baffled. I am new to
 Asterisk and there is lots to learn, but my config is pretty basic, my
 sip.conf simply has the phones and single sip provider context in it. It
 doesn't make sense that the voip provider going offline takes the whole
 setup out with it. I am suspecting something else went south at the same
 time.

 I have snom m3's and one Astra 480i.

 Thanks!
 jlc


I've seen this behaviour from Asterisk as well... while I can't say I have 
tracked it down and verified this... I've seen other talks about how Asterisk 
gets rather unhappy when it can't preform DNS queries.  I suspect that may be 
your problem.   Might want to check the archives for other issues that people 
have talked about DNS as a possible cause and see if there are any 
similarities.

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Re: [asterisk-users] Fax on FXS

2008-06-07 Thread Matt Watson
On June 7, 2008 11:37:20 am bilal ghayyad wrote:
 Hi List;

 What configuration needed to let my FXS send and
 receive FAX?


Your probably going to need to give some more details about your setup before 
anybody can help you... theres really nothing special you need to configure 
for an FXS port to attach a fax machine to it...

keep in mind that faxing over VoIP is extremely tricky at best, but if your 
entire call path is TDM then you shouldn;t have much of a problem.

-- 
Matt Watson
http://www.mattgwatson.ca

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Re: [asterisk-users] fxotune vs rxgain/txgain

2008-06-06 Thread Matt Watson

In short, fxotune adjusts line impedance, where as adjusting gains I believe 
is essentially adjusting the amplification / deamplification of the signal.

http://www.voip-info.org/wiki/view/Asterisk+fxotune

-- 
Matt Watson
http://www.mattgwatson.ca

On June 6, 2008 12:43:51 am Noah Miller wrote:
 Hi All -

 I hope somebody can clarify for me what exactly fxotune does, and how
 it is related to gain settings.  I've been reading what appears to be
 conflicting information from various sources.

 I've got a box with an AEX800 with 6 lines (from Qwest) running
 asterisk and zaptel versions 1.4.20.1 and 1.4.11 respectively.  We've
 been experiencing some echo/quality issues on certain calls which seem
 to happen on all 6 of the lines.  I manually calibrated the
 rxgain/txgain using ztmonitor and a milliwatt test line to the
 somewhat improbable levels of +10.0/-2.0 (about the same for all 6
 lines).  These settings yield acceptable call volumes, but echo and
 noise are problems.

 If I run fxotune, it gives me the following numbers:

 1=10,0,0,0,0,0,0,0,0
 2=12,0,0,0,0,0,0,0,0
 3=12,0,0,0,0,0,0,0,0
 4=10,0,0,0,0,0,0,0,0
 5=10,0,0,0,0,0,0,0,0
 6=10,0,0,0,0,0,0,0,0

 Two questions here:

 1) What do these numbers mean?  Are they in any way related to either
 rxgain or txgain?
 2) Am I supposed to set rxgain and txgain back to 0 if I use fxotune -s?

 If I do use these fxotune settings and set rxgain and txgain to zero,
 the volume on incoming zap calls is almost too low to be heard, but
 echo issues seem to be solved.

 Do I have to choose between 1) acceptable call volume with echo or 2)
 super-quiet call volume without echo?  Should I petition Qwest to
 install a repeater?


 Thanks,
 Noah

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Re: [asterisk-users] Asterisk 1.6 vs 1.4?

2008-06-04 Thread Matt Watson
Have you tuned rxgain  txgain in Zapata.conf?  shameless-self-plug 
http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/
 /plug

Also, have you used fxotune to tune each FXO interface?

I believe echo cancellation happens at the Zaptel / DAHDI level, so using 
Asterisk 1.6 probably isn't going to give you any benefit.


--
Matt Watson
http://www.mattgwatson.ca


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson
Sent: Wednesday, June 04, 2008 11:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6 vs 1.4?

Is there some location that outlines the major differences between
Asterisk version 1.4 and version 1.6?  I've read through change logs and
several technical discussions, but technical details don't really give
me the big picture.  Basically, is 1.6 more stable than 1.4?  Is it more
efficient?  Does it work better with echo cancelers like Oslec?  I'm
currently using Asterisk as a PBX for our branch offices and will soon
be converting our main office.  Our goal is to be able to have 2 analog
lines at each office, calls come in to each PBX and are routed by VOIP
to a receptionist at one of the offices who then routes calls
appropriately.  We're currently using Asterisk 1.4.19, Zaptel 1.4.10,
Oslec SVN,  Rhino R4FXO-EC cards, and Snom 300 Phones.  All of our
branch offices have 1MBPS DSL connections and are linked to each other
by VPN's running on our Cisco 1720 routers. Our only problem so far is
with intermittent echo on calls.  Most of the calls have a little echo
right at first, but it goes away almost immediately as the echo canceler
trains.  Every now and then, however, we get a call with terrible echo.
I've put in several e-mails to rhino support asking if the hardware echo
canceler needs something I haven't done but didn't get a response.  I
know echo is just something we have deal with when using analog lines,
but I didn't think it would be this big of a problem.  All of our
offices are in rural areas where digital lines are unavailable so that
is not an option.

Given this setup, is there any reason for me to switch to Asterisk 1.6
or should I stick with 1.4?

Thanks,
Brent Davidson

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Re: [asterisk-users] forwarding pots lines

2008-05-23 Thread Matt Watson
On May 23, 2008 11:25:55 am Dennis P. Clark wrote:
 Will fax and dial-up internet work through the gateway?

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Joe
 Carroll
 Sent: Friday, May 23, 2008 8:51 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
 Users Mailing List -Non-Commercial Discussion
 Subject: Re: [asterisk-users] forwarding pots lines

 There are a couple of companies out there that make 24 port fxo and fxs
 boxes. If you have some unused  fibers you cout do this very reliably
 with two channel banks...  One with fxs ports and the other with fxo
 ports and t1 media converters.

  The grand stream solution mentioned in an earlier post does 8 ports,
 you could get one 4 port model and one 8 port model of fxs and the same
 of fxo and  accomplish your goal rather inexpensively as well.


In generaly this is a bad idea (especially dialup internet). If both the 
gateways you use support T.38 origination/termination then faxing will not be 
a problem at all.

However, in your case I assume you are only transporting the calls over LAN, 
and there is no WAN/Internet involved... which means you will probably achive 
a high success rate for both dialup and fax... I wouldn;t be surprised if you 
can;t max out the baud on your dialup internet connections though... i'd 
expect a slight reduction in speed (and errors, though error correction built 
into your modem would hopefully take care of this, at the cost of a a little 
speed due to re-transmissions)



-- 
Matt
http://www.mattgwatson.ca

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Re: [asterisk-users] *#%! Polycom...

2008-05-23 Thread Matt Watson
On May 23, 2008 05:27:49 pm Ken D'Ambrosio wrote:
 I used to do lots of Asterisk, but got an offer I couldn't refuse, and
 went SysAdmin.  Well, now I'm trying to bring Asterisk in-house, and want
 to set up a test system.  One thing I'd really like to get my hands on is
 recent firmware, etc., for SoundPoint IP 430's.  Freedomphones.net, my old
 source, seems to have been kaput about as long as I've been a sysadmin;
 are there any other sources out there?  (And, yeah, if anyone wants to
 e-mail them to me directly, I won't say no.)


My source was google, and I came across this almost right away: 
http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware-polycom-0-3.0.1-2.html

presumably their RPM includes firmware for all of the polycom's

I don't use polycom's and never actually downloaded the RPM... but it seems to 
me thats what you are looking for.

You should also be able to contact whomever you bought your Polycom's from to 
obtain the most recent versions.

-- 
Matt
http://www.mattgwatson.ca

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Re: [asterisk-users] Asterisk Wackyness

2008-05-22 Thread Matt Watson
On May 22, 2008 02:06:06 pm Jared Smith wrote:
 On Thu, 2008-05-22 at 10:48 -0700, Douglas Garstang wrote:
  We didn't want to be generating actual network traffic for this, so I
  tried originating a call to [EMAIL PROTECTED]

 Why not try [EMAIL PROTECTED] and see if that solves the problem for you?  I'm
 going to make a wild guess here that Asterisk is trying to do a DNS
 lookup on whatever you have there for xxx.  Is your xxx numeric?
 I'll bet if you change it to [EMAIL PROTECTED] you won't see the problem.

specifying localhost is probably a good idea... if you just specify a random 
string say asdf more than likely whats going to happen is asterisk will try 
and do a DNS query via gethostbyname() (guessing thats what * uses).  Whats 
then going to happen is the resolver is first going to try and do a DNS query 
on asdf thats going to fail, what will happen next is it will try and 
search for asdf.your search domain specified in /etc/resolv.conf.  
Depending on what your asdf string is... that domain might actually resolv 
to something and gethostbyname() is going to return the results.

That all being said... I'm surprised you are not more concerned with fixing 
the real problem instead of your workaround...  By any chance is your SQL 
server not on the same subnet as your * box?  If not, do you have something 
like a cisco router/PIX/ASA between the subnets?  if thats the case, your 
router might be detecting the idle connection and killing it.  I know our PIX 
will do this.  I suppose even if its on the same subnet, you could have 
something running on either your * box or mysql box that will blow away idle 
connections... but that would probably be a little more obvious and you'd 
know about it.

-- 
Matt
http://www.mattgwatson.ca

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Re: [asterisk-users] Handling multiple fax machines and the fax extension, and general call routing

2008-05-22 Thread Matt Watson
On May 22, 2008 04:42:27 pm Steve Totaro wrote:

 PS.  Figured I would start with DHADI now.


psst. its DAHDI ;)




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Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)

2008-05-21 Thread Matt Watson
Does your extensions.conf have any more configuration than what you've shown?

If not, then you are lacking dialplan for anything but internal calls.

--
Matt

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNd
Sent: Wednesday, May 21, 2008 9:01 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn 
calls)

Hello all,

its been a while im trying to setup my asterisk/sipura 3102 to recieve/make 
calls from softphones on pcs in my home..
i've set up 5 SIP extensions in sip.conf and made the dialing plan in 
extensions.conf..
i could make calls from 1 sip phone to another in my home.. but i cant call out 
using pstn line interface nor recieve calls..
please find below my topology as well as config info:

 (192.168.0.0)
   LAN__
  ||   |
softphone  asterisk   sipura-PSTN LINE



Configuration:

ASTERISK:

sip.conf

[101]
type=peer
port=5062
host=dynamic
secret=1234
context=spa


[103]
type=peer
port=5061
host=dynamic
secret=1234
context=spa

[100]
type=peer
port=5061
host=dynamic
secret=1234
context=spa

[111]
type=peer
port=5060
host=dynamic
secret=1234
context=spa

== ===

EXTENSIONS.CONF

[spa]
Exten = _1XX,1,Dial(SIP/${EXTEN})

== ===


and this is the settings i have right now for sipura 3102 in my PSTN LINE:


http://img84.imageshack.us/my.php?image=40541922um2.jpghttp://www.voipuser.org/ship_to.php?url=http://img84.imageshack.us/my.php?image=40541922um2.jpg

http://img98.imageshack.us/my.php?image=55448347ss9.jpghttp://www.voipuser.org/ship_to.php?url=http://img98.imageshack.us/my.php?image=55448347ss9.jpg

http://img262.imageshack.us/my.php?imag ... 
472qz3.jpghttp://img262.imageshack.us/my.php?imag%20...%20472qz3.jpg

ps: i read so many tutorials and none seems to help..
lately whenever i try to call out using my sipphone.. it gives me 503 service 
unavailable and this is wht shows on the CLI of asterisk when i set sip debug 
on..




ubuntu-pbx-desktop*CLI
  == Connect attempt from '127.0.0.1' unable to authenticate
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/1003-b5f05600, SIP/1009) in 
new stack
-- Called 1009*CLI
-- Got SIP response 410 Gone back from 192.168.0.111
-- SIP/1009-081741d0 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/1003-b5f05600' status is 'CONGESTION'



Invite your mail contacts to join your friends list with Windows Live Spaces. 
It's easy! Try 
it!http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us
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Re: [asterisk-users] Fax Machine Options

2008-05-20 Thread Matt Watson
The first part of this is kind of off topic as it doesn't answer OP's original 
question, but instead is a reply to one of the replies.

Cisco is certainly not the only option for doing T38 gatewaying with Asterisk.

I believe Asterisk 1.6 with app_fax supports T.38 origination and termination, 
that is not gatewaying, however if origination and termination are already 
there, gatewaying should be fairly trivial to implement.  I haven't actually 
tested 1.6 using T.38, however I have read:  http://www.asterisk.org/node/48457

11873, Added core API changes to handle T.38 origination and termination
(The version of app_fax in Asterisk-addons now supports this.)

Additionally, there are some 3rd party modules available for Asterisk 1.4 that 
will add T.38 termination, origination, and gatewaying.  The ones I am thinking 
of specifically are the ones made by Attractel in there Attrafax package 
(previously known as Faxterisk): http://www.attrafax.com/attrafax.php

I have used Attrafax before and it works great for us.  We use it in 
combination with Linksys SPA2102 ATAs.  We had problems with it at first but 
upgrading the firmware on the Linksys ATAs made the problem go away.  In our 
case we have a PRI however and are not using SIP connections over the internet.

Another option as you have already stated is using a SIP provider that supports 
T.38 such as gafachi.

However in this particular case I understand the OP has already provisioned 
DIDs from a SIP provider, assuming one of these DIDs is your fax number you may 
find yourself with a bit of a problem if your provider does not support T.38.  
You may have some luck with faxing w/o T.38 using G.711a/u over the internet, 
but it will be patchy at best, you will probably find you will have many failed 
faxes doing this.  Using G.711a/u internally over a LAN is one thing (still 
wouldn't recommend it, but you would get a high success rate), but doing it 
over the internet is a completely different story.  If you have no real PSTN 
connections and are SIP only, your provider *must* support T.38 to achieve an 
acceptable success rate.

If your DIDs are already on print materials and your provider doesn't support 
T.38, the only options I would see for you are:

1) Have your Fax DID ported to another SIP provider that does support T.38, you 
can leave your voice ones with your current provider
2) Get a new DID from a another SIP provider and re-print all of your materials 
(probably incredibly expensive)

--
Matt

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem 
Helge
Sent: Monday, May 19, 2008 11:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax Machine Options

Cisco gateway with T.38 support. That's the only real way to do faxing
through asterisk. I think a VG200 with newer firmware will support SIP
+  T.38 but don't buy on my suggestion because I've never used that
device outside call manager configuration.

Or see if your VoIP provider supports T.38 fax but you must use SIP in
that case. It will work very well once you get it working hint:
check udptl.conf



On Mon, May 19, 2008 at 11:27 PM, Joseph L. Casale
[EMAIL PROTECTED] wrote:
 Is my only solution to add a fax machine to our VOIP only setup by using an
 IAXy?
 I should specify the office people want a traditional fax machine in the
 sense that

 fax's be sent and received from a physical unit, they don't want an email to
 fax setup.
 They have a dedicated sip did provisioned just for the fax.



 What are others using?



 Thanks!
 jlc

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Re: [asterisk-users] Fax Machine Options

2008-05-20 Thread Matt Watson
Interesting...  I really don;t know the T.38 protocol other than what it does.  
How it goes about doing it I haven;t really gotten into.

I would of thought that gatewaying would of (essentially) be a bridge between a 
termination and origination action.  However that is just completely sort of 
what i think without any real evidence behind my thoughts.

One day i'll have to read up on it.

--
Matt

From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Steve Underwood [EMAIL 
PROTECTED]
Sent: Tuesday, May 20, 2008 9:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax Machine Options

Matt Watson wrote:
 I believe Asterisk 1.6 with app_fax supports T.38 origination and 
 termination, that is not gatewaying, however if origination and termination 
 are already there, gatewaying should be fairly trivial to implement.  I 
 haven't actually tested 1.6 using T.38, however I have read:  
 http://www.asterisk.org/node/48457


T.38 gateway is a totally different problem than T.38
origination/termination. They share very little code, and almost none of
their design.

Regards,
Steve


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Re: [asterisk-users] Server recommendation help

2008-05-20 Thread Matt Watson
You might want to see if you can change the IRQ assignments in your servers 
bios (might have to turn off the PNP OS Installed option if you have one)

--
Matt

From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Cavanna, Richard [EMAIL 
PROTECTED]
Sent: Tuesday, May 20, 2008 11:04 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Server recommendation help

On Tue, May 20, 2008 at 07:55:32AM -0500, Cavanna, Richard wrote:
 I am having a issues with adding a analog card to my dell 2800.  I
 already have a t1 card installed and running fine but when I install
the
 analog card asterisk will not start (ztcfg fails).  I have determined
it
 is because of a IRQ problem and have decided to get a new server.  Can
 anyone suggest a server grade setup that supports this?  I would
rather
 not buy a machine that will be unstable.  I am not opposed to building
 one if need be.

You omit some obvious details, such as the actual error message you get,
your /etc/zaptel.conf and the output of 'cat /proc/zaptel/*' .

I figure that the driver for the analog card loads before the driver for
the T1 card, hence invalidating your configuration.



 Cards I have installed:
 DigiumTE205P - 5v
   TDM410

Thanks for the offer to help trouble shoot but I never even got that
far.  On boot it actually makes my Ethernet ports fail.  I am thinking
of disabling the internal ports and adding in a card.  Might help but as
this box is in production I do not have the ability to do that much
testing on it. I am looking at some planned downtime next weekend to go
further with this.

Also this box is ~3 years old so I figure If I can find a box that is
not too expensive I can migrate this to a FTP box and get a new *
server. As added bonus I can use the reload to clean up my dial plan.

If anyone has a TE205P-5v and TDM410 running is same box I would like to
know the setup.

Richard


--
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Re: [asterisk-users] BLF Compatible Phones

2008-05-19 Thread Matt Watson
On May 19, 2008 12:51:09 pm Grygoriy Dobrovolskyy wrote:
 Hmm, i dont like aastra really much, their transfer management is not human
 friendly ;)

What do you mean by that?  I've run my Aastra's with BLF using both 
Aastra's 'blf' mode and 'blfxfer' mode... the former is basically attended 
transfer, and the later is basically blind transfer... both of which our 
staff haven;t had any problems with.

using 'blf' its basically
1. press xfer
2. press blf key
3. ask if they want to take the call
4. press xfer again

using 'blfxfer' its:
1. press blf key 

In both modes the blf keys also act as speed-dial keys.

I don't really see how either of those is not 'human friendly'?  Or maybe I'm 
just completely off on what you mean.


-- 
Matt
http://www.mattgwatson.ca

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Re: [asterisk-users] Asterisk first time user

2008-05-19 Thread Matt Watson
On May 19, 2008 03:21:34 pm Aaron Stranberg wrote:
 Folks,
   We are a small office with remote users less than 20 total phone
 extensions, and I am looking for some guidance on choosing between
 asterisknow and a centos/ubuntu or any other os with an asterisk +
 asteriskgui build out?  Looking to get up and going quick with some method
 of GUI administration that won't require a ton of ongoing linux admin level
 support.  I hit a couple of stumbles going the asterisk + asterisk GUI
 route (404 errors on ivr page etc..)  and am tempted to take the easy path
 of asterisknow iso and go.  Thanks for any pointers, and advance apologies
 if this had been beat to death.

 -Aaron

IMHO, there is really no way to say this one is best.  Each solution might 
be better at X while the other is better at Y... its very dependent on your 
situation

Though, I gather you'd rather not deal with the actual OS-level, so you are 
probably best to stick with one of the complete packages like AsteriskNOW, 
Trixbox (they have a free and paid version), PBX in a Flash, and i;m sure 
there are many others... 

I haven't used any of them however so I can't really speak about the pros and 
cons of them.

-- 
Matt
http://www.mattgwatson.ca

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Re: [asterisk-users] Fedora 9 + Asterisk

2008-05-19 Thread Matt Watson
On May 19, 2008 06:49:23 pm Kevin Smith wrote:
 I almost hate to admit this...but I'm still running Asterisk 1.2 on
 Fedora 4 :D

IMO theres nothing wrong with running an old version of * or an old version of 
the OS... as long as the box doesn't have a public IP bound to it.

If its an internal-only box and its running bug-free, why screw with it?  If 
you can get it to over the internet however... security becomes a big 
concern.


-- 
Matt
http://www.mattgwatson.ca

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Re: [asterisk-users] Digium Card: Power Connector, from SATA to NORMAL

2008-05-17 Thread Matt Watson
On May 17, 2008 06:59:43 am Gordon Henderson wrote:
 On Sat, 17 May 2008, bilal ghayyad wrote:
  Well, why Digium is still using this kind of power
  connector while all new machines does not come with
  these types?

 The new machines that I buy come with legacy power connectors. The flash
 IDE drives I buy need legacy power connectors, and since convertors are
 trivially avalable why is it an issue?


molex power connectors are not legacy or old style except when used in 
reference to SATA devices.  Seeing as how Digium interface cards are not SATA 
devices, why would you expect them to be using a SATA connector?  Its not 
Digiums fault that the PSU you bought doesn't include molex connectors.

SATA uses a different power connector for a few reasons, but the biggest is 
that SATA supports hot-plugging (assuming your controller, drive, and OS 
support it), in order for hot-plug to work the drive needs a 3.3V voltage as 
well as  5V and 12V, molex only gives 5V and 12V.  The actual physical 
connector that molex uses also does not lend itself very well to 
hot-plugging.

SATA power connectors, while there is no real reason that non-SATA devices 
couldn't use them, they simply were designed for SATA specifically.  

Personally, I prefer molex connectors for most things simply because they are 
far more secure than SATA connectors (at least the ones i've used).


-- 
Matt
http://www.mattgwatson.ca

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Re: [asterisk-users] Asterisk concurrent calls count

2008-05-16 Thread Matt Watson
You can NOT use bogomips as any kind of measurement for system performance.

First of all, Bogomips is a linux-specific thing and not available on other OS 
that Asterisk runs on.

The second, and far more important point.  Bogo is taken from the word 
Bogus.  Bogomips are not a measurement of system performance, it is simply a 
number used for calibrating parts of the kernel for your CPU.

The problem with coming up with these numbers of concurrent calls is that 
Asterisk is not a complete package.  Meaning, it's the software portion only, 
most other systems when you get them are going to be the software  the 
hardware in one package, the 2 go hand in hand and are specifically designed 
for each other.

Asterisk does not fall into that category unless you invest in one of the many 
asterisk appliances out there.  Digium has no control over what hardware you 
are going to run Asterisk on, so they can't provide you with these numbers.

Heres a few questions at the top of my head that I think would influence the 
answer:

are you recording calls? are you transcoding calls?  are you using T1s or 
SIP/IAX trunks? Did you buy the 7.2krpm, 10krpm, or 15krpm hard-drives?  Do 
your harddrives have 8mb, 16mb, or 32mb cache? Did you buy the better SAS 
controller?  Did you buy 667mhz or 800mhz ram? Are you using EXT3, ReisferFS, 
XFS, JFS, ZFS, UFS?  Are you using AGIs?  Are you using MeetMEs? How many?  
Whats the average length of the conferences?  Are devices using re-invites to 
take Asterisk out of the call loop?

The list goes on and on... and every single one of those answers is going to 
influence that number for How many calls can my system handle?

--
Matt

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Backeberg
Sent: Friday, May 16, 2008 9:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk concurrent calls count

I wonder if there's a proportion where somebody could take some
standard kernel output, say bogomips,
and guesstimate some proportionality from that. As in: bogomips says
this, expect ballpark 120 SIP over codec calls.
It certainly seems like there could be some kind of asterisk
benchmarking utility kindof like Sandra for Windows. I know there are
a gazillion variables in asterisk, and that's why asterisk is so
powerful. But some benchmarking utility would at least allow some
(even if phony baloney) relative comparisons between similar hardware.

Has anybody ever tried to roll their own VoIP or Zaptel load
simulator? How did they do it?

On Fri, May 16, 2008 at 7:59 AM, Al Baker [EMAIL PROTECTED] wrote:
 this is one very weak area for *. There is NO ANSWER.
 Now in fairness to *, the answer DOES depend on a # of critical variables.
 How much CODEC to CODEC transcription is going on.
 How many MEET Me conferences are going on.

 On the other hand, DIGIUM COULD, since they have a lab take 4-5
 'standard' workloads
 on two of the most common hardware boxes, say Dell  HP, and run x # of
 transcriptions and
 show the #'s.
 Then x # of meet-me conferences.

 Face it the DB Industry did this 15-2- YEARS ago with TP benckmarks

 Rockwell and NORTEL can tell you this for every piece of hardware they sell.

 It is a an area DIGIUM need to man-up in.

 Alexey Shimeshov wrote:
 Hello, Alexander.

 AO Hi Asterisk Users,

 AO I'm interested in how many concurrent calls Asterisk can process without
 AO troubles. I mean 1 Asterisk server (software) like either proxy or media
 AO server (any numbers will be appropriate).

 AO 1. Is there any limitations by the software? What is this number?
 AO 2. What is the maximum count of concurrent calls you've ever seen/tested?

 Look at this example

 http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm



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Re: [asterisk-users] Asterisk for Larg

2008-05-15 Thread Matt Watson
You'd probably want to run something else to handle your registrations like 
OpenSER with that many phones.

--
Matt

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bhrugu Mehta
Sent: Thursday, May 15, 2008 8:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk for Larg

hi,
I have not tested that but I have seen 100 agents configure with asterisk.
thnks
Bhrugu mehta

On 5/15/08, gmail [EMAIL PROTECTED] wrote:


 Is Asterisk practically stable and reliable for a larg Enterprise has say a
 1 phones , is there any case study like this?
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Re: [asterisk-users] *72 Telco Call Forwarding

2008-05-15 Thread Matt Watson
Is there any reason you don't want to use Wait()?

However, I would use WaitForRing() myself - its also a great solution on dirty 
analog lines where you receive phantom calls.

That being said, I don't know how to do it without using some form of Wait.. as 
far as I know zapata.conf doesn't provide a method of telling Asterisk to wait 
for a specific period of time or rings
--
Matt

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann
Sent: Thursday, May 15, 2008 12:07 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] *72 Telco Call Forwarding

Is there a way to force asterisk to ignore the first ring of a call without 
using Wait() ?

When I active *72 call forward on my analog lines from the telco, they always 
send a single ring and then do the forwarding.  Asterisk picks up essentially a 
dead line and rings the phones which gets really annoying.

Thanks.


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Re: [asterisk-users] Call only for registered sip users...

2008-05-13 Thread Matt Watson
Do you mean

What do I need to configure on my * installation so that only registered sip 
users can make calls?  ?

If so, you are going to need to give a lot more details regarding your current 
configuration for you to get any answers.

--
Matt



From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software
Sent: Tuesday, May 13, 2008 2:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call only for registered sip users...

What I need to configure in my * to permit make calls only registered sip 
users??

Thanks!
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Re: [asterisk-users] BLF Compatible Phones

2008-05-13 Thread Matt Watson
I'm using Aastra 57i + 560M sidecars for receptionists... the only downside is 
that they support a max of 50 BLF subscriptions... you can setup up to 180 blf 
keys with 3 560Ms but it will still only subscribe to a max of 50... from what 
I understand it's a firmware limitation.

For 4-6 phones you could probably get away with doing it directly on the 57i 
with no 560M's (or 536M's) too many more phones and you'd need the sidecars 
just for the extra buttons I think.

--
Matt

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Tuesday, May 13, 2008 3:50 PM
To: asterisk-users@lists.digium.com; asterisk-users@lists.digium.com
Subject: [asterisk-users] BLF Compatible Phones


I am new to asterisk and am looking to setup a small office with 4-6 IP
phones and 4 analog lines from the local telco (primary line with HUNT
to the other lines). I am considering purchase of a Digium AEX800.

One of the features that will be important (particularly for the
receptionist desk is to show status of the other lines in use). I don't
want the receptionist to pick up a line if it being used.

Is there a list of phones that are BLF (Busy Lamp Field) compatible? I'm
assuming (after reading tons of misc articles) that this is what I need
in order for the receptionist not to pick up lines in use. If this is
not the case please set me straight.

I am considering the cisco 7960's, linksys SPA942, and possibly some
polycom phones. I was leaning toward the 7960 but I've read that it is
not BLF compatible. Are there any workarounds for this? I am new to the
game and would be grateful for any recommendations on which phones would
be the easiest to setup, etc. I currently have a working asterisk
install at home with a single cisco 7960 registered which isn't hooked
up to any trunks as of yet.

Thanks,

Dayton Gray

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Re: [asterisk-users] 3U server chassis Digium TE405P?

2008-05-12 Thread Matt Watson
I'm not sure if a full-height card would fit (vertically) in a 3U chassis... 
but I would probably also assume that if it would not, that the chassis/mobo 
would have a PCI/PCI-Express riser card that would mount the cards horizontally.

Might want to check that out with the manufacturer of the chassis.

--
Matt

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan
Sent: Monday, May 12, 2008 11:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 3U server chassis  Digium TE405P?

Gentlemen,

First let me say it's great to be back on the Asterisk mailing lists.
Those of you who have been around for a while will remember me as
Rushowr. I look forward to answering questions and whatnot in the
future, but for the moment I have a minor question that I cannot find a
definitive answer for online.

I am in possession of a Digium TE405P card which I _know_ will fit in a
4U chassis, but we are building a new server and cannot get a 4U from
the supplier that my current client wants to use. However, we can get a
3U chassis. My question is, will this card fit? Does anyone out there
have a 405 out there that they have installed in a 3U?

Thanks in advance for any help that can be offered,
Sherwood McGowan
VoIP / Telecom Solutions Consultant

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Re: [asterisk-users] Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect?

2008-05-11 Thread Matt Watson
FreePBX has this functionality... they call it Confirm Calls

I;m not sure if you can set it on actual extensions, but I know you can set it 
on ring groups.

I don't imagine the dialplan for doing it is very complicated if you wanted to 
do it by hand.

--
Matt

From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Robert DeVries [EMAIL 
PROTECTED]
Sent: Sunday, May 11, 2008 12:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Anyone Know How to Have Asterisk Work Like
GranCentral and Require a Touch-Tone to Connect?

GrandCentral has a feature where when you call the GrandCentral number it can 
ring multiple phones.  However, it's not the first phone to answer that gets 
connected, but the first phone to answer AND play a touch-tone after hearing a 
recording.  The advantage of this is that if one of the called phones has 
voicemail, it won't get connected to the calling party because the VM won't 
send a touch tone in response to the recording, unlike a live person.

I have always resisted implementing a multiple ring scenario with Asterisk that 
included a cellphone because of the voicemail answering problem, but this seems 
to be a solution.

Anyone know how to implement it with Asterisk?

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Re: [asterisk-users] Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect?

2008-05-11 Thread Matt Watson
I just took a quick look at the dialplan that freepbx uses for doing call 
confirmation... the dialplan part of it is actually quite simple... its just a 
matter of setting the USE_CONFIRMATION varialbe =TRUE.

However, the actual magic looks like it happenes through its dialparties.agi... 
which is a little more complicated than i'd like to try and dissect on a sunday 
afternoon!

but that might be a good place to look at how its done to learn by example.

I know in the freepbx implementation what it does is whenever a handset thats 
part of the ringgroup answers, they get a recorded message You have an 
incoming call, press 1 to accept maybe it says something else too... can;t 
recall at the moment.  The first member of the Ring group to hit 1 gets the 
call... if more than 1 person picks up the handset right away, the first to hit 
1 gets it, and the rest hear a sorry, too late, somebody else got it-type 
message (no idea what it actually says).



From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Steve Totaro [EMAIL 
PROTECTED]
Sent: Sunday, May 11, 2008 1:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Anyone Know How to Have Asterisk Work Like
GranCentral and Require a Touch-Tone to Connect?

On Sun, May 11, 2008 at 12:24 PM, Robert DeVries [EMAIL PROTECTED] wrote:
 GrandCentral has a feature where when you call the GrandCentral number it
 can ring multiple phones.  However, it's not the first phone to answer that
 gets connected, but the first phone to answer AND play a touch-tone after
 hearing a recording.  The advantage of this is that if one of the called
 phones has voicemail, it won't get connected to the calling party because
 the VM won't send a touch tone in response to the recording, unlike a live
 person.

 I have always resisted implementing a multiple ring scenario with Asterisk
 that included a cellphone because of the voicemail answering problem, but
 this seems to be a solution.

 Anyone know how to implement it with Asterisk?


GREAT IDEA!  (even if it wasn't yours ;-)

I have had so many issues with this and desk phones, cell phones being
out of range, turned off, or answering machines set to answer after
two rings.

If this gets implemented, it would be a great feature and save me tons
of complaints and explanations.

Maybe a posting on the dev list is appropriate.  I would certainly
contribute to a bounty.

Thanks,
Steve Totaro

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[asterisk-users] Digium AEX410

2008-05-11 Thread Matt Watson
Poking around the zaptel SVN earlier today i see support was added for an 
AEX410 card recently...

I'm going to go out on a limb and assume this is the PCI-Express version of the 
TDM410?

Any hints on a general availability date?

--
Matt

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Re: [asterisk-users] Best Linux distribution to use in Asterisk server

2008-05-10 Thread Matt Watson
I'm a Gentoo guy myself too... but the best advice I can give is just 
re-hashing what others have already set... pick whatever you are the most 
comfortable with... and if support contracts are important to you, then that 
will be a factor as well.  I've used most of bigger distros out there over the 
last 10 years, but right now Gentoo is where I am at.

It also might depend on if you intend on using Asterisk from the package system 
fo your distro or if you intend on compiling it yourself.  On my * box I 
compile Asterisk, Zaptel, LibPRI by hand, everything else I've installed from 
portgage (Gentoo's package system).

--
Matt

From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Marco [EMAIL PROTECTED]
Sent: Saturday, May 10, 2008 4:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best Linux distribution to use in Asterisk
server

Personally, I love the debian way, but I must admit that when it gets to 
Asterisk, I prefer to use a RedHat-based distro like CentOS, first of all for 
the proven reliability, then for the widely used rpm packaging system and last 
because there are many distro CentOS-based that provide a stable system with 
FreePBX and all the stuff :-P

Hi, I allways use Gentoo y my Asterisk servers and work well, but what do
you think about to use Ubuntu or another distibution??

Thanks

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Re: [asterisk-users] better enumlookup handler

2008-05-07 Thread Matt Watson
There is a enumlookup.agi that is included with FreePBX and thus trixbox, PBX 
in a flash, etc. etc.

If you have trouble finding it let me know and I can send you it.

I can;t really vouch for its quality, but I do use it and it does work... but 
i;m not sure how well it handles multiple results.  I know it will successfully 
connect to systems that give multiple results, i;m just not sure if it does 
infact failover if the first one doesn;t work.

--
Matt

From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Brian J. Murrell [EMAIL 
PROTECTED]
Sent: Tuesday, May 06, 2008 10:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] better enumlookup handler

Does anyone have a better ENUM lookup handler than the built-in
ENUMLOOKUP() function?  The built-in function does not properly handle
multiple return values such as:

8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 u E2U+SIP 
!^\\+1866(.*)$!sip:[EMAIL PROTECTED] .
8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 u E2U+SIP 
!^\\+1866(.*)$!sip:[EMAIL PROTECTED] .

And thus does not handle roll-over should one be unavailable for
whatever reason.

There is this voip-info.org wiki page:
http://www.voip-info.org/tiki-index.php?page=Asterisk+and+multiple+ENUM
+entries but the downloads that it's pointing to seem to be dead.

Sure I could take to writing an AGI script and probably be done it in a
few hours, but why re-invent the wheel?

Thanx,
b.


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Re: [asterisk-users] Setting the TOS using IPtables screws up the DSCP field

2008-05-07 Thread Matt Watson
Why are you trying to change the ToS from 46 (0x2e) Expedited for the RTP/RTCP 
packets to 16 (0x10)?

I mean... these values really only need to be meaningful to yourself, your 
switches, your routers etc however

ToS 46 (0x2e) is the standard value for RTP / RTCP as it is basically the 
highest priority value you can assign to something... ToS 46 is basically 
suppose to indicate that it trumps all other traffic and should be send before 
anything else (Which is a good thing for the RTP traffic)

The SIP Signalling traffic is a little less important and its standard ToS 
value is 26 (0x1a).

You also don;t need to use IPTables to set these values... Asterisk will do it 
for you as long as you have installed libcaps (I believe its required for it).

And I don;t know what phones you are using... but your phones are probably also 
setting these values for you I know the Aastra phones have QoS/ToS settings 
under Options - Network - Type of Service

--
Matt

From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Vikas [EMAIL PROTECTED]
Sent: Wednesday, May 07, 2008 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Setting the TOS using IPtables screws up the DSCP 
field

Concise summary: When I set the TOS to Minimize-Delay the DSCP field
in the packet changes from Expedited Forwarding to Unknown

Here are the details:

Scenario 1: IpTables is not used to set the TOS

This is what the packet looks like using wireshark:
Internet Protocol, Src: 59.93.192.xx (59.93.192.xx), Dst:
64.62.134.yyy (64.62.134.yyy)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0xb8 (DSCP 0x2e: Expedited
Forwarding; ECN: 0x00)
1011 10.. = Differentiated Services Codepoint: Expedited
Forwarding (0x2e)
 ..0. = ECN-Capable Transport (ECT): 0
 ...0 = ECN-CE: 0


Scenario 2: IpTables is used to set the TOS

Output of $/etc/rc.d/init.d/iptables status
Table: mangle
Chain PREROUTING (policy ACCEPT)
num  target prot opt source   destination

Chain INPUT (policy ACCEPT)
num  target prot opt source   destination

Chain FORWARD (policy ACCEPT)
num  target prot opt source   destination
1TOSudp  --  0.0.0.0/00.0.0.0/0   udp
dpts:5060:5069 TOS set 0x10
2TOStcp  --  0.0.0.0/00.0.0.0/0   tcp
dpts:5060:5069 TOS set 0x10
3TOSudp  --  0.0.0.0/00.0.0.0/0   udp
dpts:1:2 TOS set 0x10

This is what the packet looks like using wireshark:
Internet Protocol, Src: 59.93.192.xx (59.93.192.xx), Dst:
64.62.134.xxx (64.62.134.xxx)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0xb0 (DSCP 0x2c: Unknown DSCP; ECN: 0x00)
1011 00.. = Differentiated Services Codepoint: Unknown (0x2c)
 ..0. = ECN-Capable Transport (ECT): 0
 ...0 = ECN-CE: 0


I have no idea what am I doing wrong.

Here is some reference reading I did:
http://www.tucny.com/dscptos

Any pointers in the right direction will be very much appreciated.

Thanks for your time,

Sysadmin
http://www.debtconsolidationcare.com
Internets First get out of debt community

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Re: [asterisk-users] dundi network - redundancy / fault tolerance ?

2008-05-07 Thread Matt Watson
I don;t have any answers for you...

But I would love to hear about the results after you get this working and what 
road blocks you hit and how you overcame them.

--
Matt

From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Ex Vito [EMAIL PROTECTED]
Sent: Wednesday, May 07, 2008 10:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] dundi network - redundancy / fault tolerance ?

  Hi list,

  I'm planning a private DUNDi network for a cross-country
  distributed PBX. Initially it will be composed of about 10
  systems, growing to about 20.

  Current requirements point to a topology of two interconnected
  DUNDi hubs, each peering with half the PBXs... This would
  lead to two interconnected / inter-peered stars.

  Example:

  - Consider PBXs A to H
  - C and E will be hubs and peer with each other
  - A, B and D peer with C
  - F, G and H peer with E

  This leads to a maximum three hop lookup and will make
  good use of current network topology / bandwidths. Of course,
  should any of the hubs be unavailable and the lookup capability
  is severely compromised.

  Now, how to move on to acheive some kind of fault tolerance ?
  According to the docs we've studied, DUNDi does not like loops
  (which we assume one can limit with low enough TTLs).

  Our doubts are:

  - Should one use the order peer parameter to specify alternate
lookup paths / peers ? Is that its purpose ? If not, what is it used
for ?

  - Alternatively, should one create loops in the DUNDi topology and
limit them via TTL ?

  - If both options are possible, which would be the trade-offs between
them ? (Not clear at all to us!)

  - Assuming any of the above is possible as a means to acheive
redundancy, which of the following topologies would your prefer ?
(hmmm, maybe I need to refresh my graph theory...) ;-)

#1 - Peer each PBX with both hubs
#2 - Duplicate both hubs and peer each PBX with its hub and
  its hub dup

For better understanding, take a look at:

#1 - http://www.2photosharing.com/images/qhpnzycd7j7kf26j2f.png
#2 - http://www.2photosharing.com/images/npzbwvgnr4t079laou0.png

  Thanks in advance for review and feedback.
  Cheers,
--
  exvito

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Re: [asterisk-users] DUNDi call impossible in one direction

2008-05-07 Thread Matt Watson
Are you using IAX2 as your transport between the 2 servers or SIP?

If you are using IAX2, are you using Asterisk 1.4.18.1 or 1.4.19.1 on either 
machine?  If so, you may be encountering the IAX2 bug that some have been 
discussing on the list recently you can read it here: 
http://lists.digium.com/pipermail/asterisk-users/2008-May/211000.html

--
Matt

From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Russell Bryant [EMAIL 
PROTECTED]
Sent: Wednesday, May 07, 2008 6:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi call impossible in one direction

Andrea Spadaccini wrote:
 I've set up DUNDi between two asterisk boxes, and sometimes happens that calls
 from machine A can't reach peers in machine B, but calls from B to A work
 correctly.

 The strange thing is that the CLI command 'dundi show peers' shows correctly
 the registered peer in both servers, and in this situation if I make a call
 from B to A, suddenly peers in server A are able to call peers in machine B.

Try using the DUNDi query CLI command to see what results your server is getting
when you try to make calls.

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

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Re: [asterisk-users] Asterisk in Production ?

2008-05-06 Thread Matt Watson
I'm using 1.4.18 in production on 2 boxes... one of which being a custom built 
desktop basically, the other being a Dell 1950 III

We are in a migration phase to the Dell box, right now the 1st box is doing 
nothing more than being a PSTN gateway to some FXO lines... basically waiting 
for numbers to be ported off the analog lines and onto the new T1 which is 
connected to the Dell box.

We have the 2 boxes connected by IAX2 trunk.

I had 1.4.19 and 1.4.19.1 running on the Dell box, but it started giving me a 
lot of trouble with the IAX2 trunk, the trunk would (seemingly) go into 
UNREACHABLE status and never come back without restarting asterisk (reload, or 
iax2 reload wouldn’t cut it).  Also, occasionally people trying to make 
outbound calls (and this probably happened on inbound as well), would get a 
all circuits are busy message because of the IAX2 channel driver reporting 
congestion on the trunk even though it was up (and not congested)

Unfortunately as this is a production box I didn’t really have time to try and 
debug it so I simply downgraded to .18 since it has proven itself well on the 
1st box.  So far since I;ve downgraded to .18 I haven’t had any problems.

Both installs I have running ontop of Gentoo (wouldn’t recommend it if you are 
new to Linux or don’t like tweak-ability).

That all being said, I'll probably give .20 a try when its released, as I see 
there have been some IAX2 bug fixes in it... but also by the time .20 is 
released I probably will have retired the box being used as a PSTN gateway and 
won’t need the IAX2 trunk anymore.

--
Matt

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vinícius Fontes
Sent: Tuesday, May 06, 2008 8:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk in Production ?

There were some really unstable Asterisk releases in the 1.4 branch. I 
personally use 1.4.13 or 1.4.15 in production. Every single time I tried 1.4.16 
or higher I had problems.



Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.

- Steve Totaro [EMAIL PROTECTED] escreveu:

 On Tue, May 6, 2008 at 7:38 AM, Benoit Plessis [EMAIL PROTECTED]
 wrote:
 
   Hi,
 
   I'm wondering what version of asterisk people use in production
   environnement ?
   on which distribution ?
 
   And what is your setup like ?
 
   We are actually running an AsteriskNow appliance with asterisk
 1.4.18.1
   and it's quite unstable.
   We have ~30 IAX2 SoftPhones and encounter some Avoiding IAX
 destroy
   deadlock
   and now that we have added a Queue, it's worse than ever. The queue
 goes
   stuck quite often
   (agent are stuck in 'In use' state and if they logoff they can't
 log-in
   till an asterisk restart).
 
 
   regards
 

 I am personally a proponent of Asterisk 1.2.X as I see more and more
 fatal bugs in the 1.4.X code come up on the lists as well as IAX2
 bugs.  I constantly hear Asterisk 1.4.whatever is much better, but
 the bugs coming out are not just unexpected behavior that one could
 live with, they are segfaults, system crashes, modules not getting
 installed (Zaptel).

 I use SIP since I have seen quite a few issues with IAX2 that were
 solved by simply switching to SIP.

 The above two yield solid systems under heavy load for me.  OS is not
 so important I do not believe.  I have some running FC8 and more
 running CentOS, both rock solid.  I think the general consensus on OS
 is use what you are most familiar with.

 While these may not be popular opinions, I still ask, what does
 SwitchVox use?  What do some of the guys around here that setup large
 systems use?  Is ABE even using 1.4 yet?  All I see in the ABE
 release
 notes is 1.2 although I have heard that ABE should be running 1.4
 Very Soon many many moons ago
 http://www.digium.com/en/docs/ABE/README .  So either Digium doesn't
 trust 1.4 enough to use it for ABE or the README is out of date.

 Thanks,
 Steve Totaro

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Re: [asterisk-users] Performance issues

2008-05-06 Thread Matt Watson
Google is awesome

http://www.voip-info.org/wiki-Asterisk+AGI

--
Matt

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chetherston miles
Sent: Tuesday, May 06, 2008 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Performance issues

Hello,

We are thinking in use asterisk-java to an billing solution, wich is the better 
choice, and if someone could give us a understandable description about the 
difference between DeadAGI and FastAGI, i found a very interesting project  
called asterisk2billing and they use DeadAGI, anyway wich one scale better?

And there is a tool for test performance?

Thanks,
Roberto
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[asterisk-users] PRI D-Channel reconfiguration = crash asterisk?

2008-05-06 Thread Matt Watson
Hello,

I just had to have MTS Allstream fix a new T1 install that we have that we 
aren't running in production yet, but it is attached to a production  machine.

Apparently they setup the T1 with only a 1 B-channel (how useful!)  even though 
we had ordered it fully loaded with 23.  Anyways... they just reconfigured the 
T1 to activate all the T1 channels and this is what I got on my * console:

  == Primary D-Channel on span 1 down
nelson*CLI
Disconnected from Asterisk server

^^ asterisk crashed.

Unfortunately I didn't have * setup on this box to dump a core file, so the 
only additional debug info I can provide is from my asterisk log file:

 [May  6 11:42:23] VERBOSE[16656] logger.c:   == Primary D-Channel on span 1 
down
[May  6 11:42:23] WARNING[16656] chan_zap.c: No D-channels available!  Using 
Primary channel 24 as D-channel anyway!
[May  6 11:42:23] WARNING[16656] chan_zap.c: The PRI Call have not been 
destroyed

Those are they only 3 relevant lines in the log file.


--
Matt

Disclaimer Statement: This e-mail is confidential and is intended for the 
above-named recipient(s) only. If you are not the intended recipient and/or 
have received this e-mail in error, please notify us by telephone and delete 
this e-mail from your system without retaining a copy in any form. Any 
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Re: [asterisk-users] PRI D-Channel reconfiguration = crash asterisk?

2008-05-06 Thread Matt Watson
My bad, I also should of mentioned...

That was on Asterisk 1.4.18 and Zaptel 1.4.10

Using a TE220B

--
Matt

From: Matt Watson
Sent: Tuesday, May 06, 2008 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: PRI D-Channel reconfiguration = crash asterisk?

Hello,

I just had to have MTS Allstream fix a new T1 install that we have that we 
aren't running in production yet, but it is attached to a production  machine.

Apparently they setup the T1 with only a 1 B-channel (how useful!)  even though 
we had ordered it fully loaded with 23.  Anyways... they just reconfigured the 
T1 to activate all the T1 channels and this is what I got on my * console:

  == Primary D-Channel on span 1 down
nelson*CLI
Disconnected from Asterisk server

^^ asterisk crashed.

Unfortunately I didn't have * setup on this box to dump a core file, so the 
only additional debug info I can provide is from my asterisk log file:

 [May  6 11:42:23] VERBOSE[16656] logger.c:   == Primary D-Channel on span 1 
down
[May  6 11:42:23] WARNING[16656] chan_zap.c: No D-channels available!  Using 
Primary channel 24 as D-channel anyway!
[May  6 11:42:23] WARNING[16656] chan_zap.c: The PRI Call have not been 
destroyed

Those are they only 3 relevant lines in the log file.


--
Matt

Disclaimer Statement: This e-mail is confidential and is intended for the 
above-named recipient(s) only. If you are not the intended recipient and/or 
have received this e-mail in error, please notify us by telephone and delete 
this e-mail from your system without retaining a copy in any form. Any 
unauthorized use or disclosure of this e-mail is prohibited.

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Re: [asterisk-users] This e-mail is confidential ... (was: Re: PRI D-Channel reconfiguration = crash asterisk?)

2008-05-06 Thread Matt Watson
That's fine... honestly I hate the message myself, however corporate policy is 
corporate policy so there isn't much of a point in discussing it.

That being said, the message does clearly say that the message is for the named 
recipients, in this particular case, the named recipient is a public mailing 
list.  By my action of sending a message to a public mailing list, one can say 
there is implied consent that it gets distributed to whomever the mailing list 
chooses on my behalf.

Thanks,

--
Matt

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
Sent: Tuesday, May 06, 2008 12:27 PM
To: Asterisk Users
Subject: [asterisk-users] This e-mail is confidential ... (was: Re: PRI 
D-Channel reconfiguration = crash asterisk?)

Matt Watson schrieb:

 Disclaimer Statement: This e-mail is confidential and is intended for the 
 above-named recipient(s) only. If you are not the intended recipient and/or 
 have received this e-mail in error, please notify us by telephone and delete 
 this e-mail from your system without retaining a copy in any form. Any 
 unauthorized use or disclosure of this e-mail is prohibited.

Your confidential e-mail is going to end up on Google ...

Regards,
  Philipp Kempgen

--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? - http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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Re: [asterisk-users] Storing voicemail on samba share

2008-05-06 Thread Matt Watson
It would probably be wiser to run an IMAP server and do imap storage instead of 
writing to a cifs-mounted directory... or use ODBC storage... assuming they are 
running a database server somewhere.

I don't have any experience with having * write voicemail files to CIFS/SMBFS, 
but I also think its not something I would try... I've personally always found 
that most network file systems don't tend to handle disconnects (server 
reboots, network outages, etc.) very well.  Mind you, it might of come along 
way since the last time I tried.

--
Matt

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical 
Support
Sent: Tuesday, May 06, 2008 3:34 PM
To: 'Asterisk Users List'
Subject: [asterisk-users] Storing voicemail on samba share

A client has asked that our asterisk installation leverage their large 
investment in their existing data center infrastructure.  We're thinking about 
putting the voicemail messages onto a Samba share (on their file servers).  Any 
pros/cons to this?  Does network/samba latency create choppiness?

Thanks,
MD
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Re: [asterisk-users] China vaults past USA in Internet users - now 220 million users in China

2008-05-04 Thread Matt Watson
What Godaddy.com has told you is more or less correct.

Its not their fault that Chinese visitors cannot hit your pages... the internet 
is China is highly censored, and quite often they firewall even very large big 
name sites like BBC news.  Typically they block sites that have any type 
discuss any type of political matters that might be going on in China, or blog 
sites where chinese citizens might speak out.  I'm not saying your site is 
one of these, but if they are infact doing it by IP address, its perfectly 
possible that your site just happens to be hosted on the same IP (or even IP 
block) as a site they decided to firewall - or perhaps a site used to occupy 
the same address space as you and they just haven't noticed its no longer there 
and un-firewalled it.

And yes, godaddy.com cannot guarentee that if you change IP addresses that the 
new one will work... just like they can't guarentee that myself or any visitors 
to my home will be able to access your website from my internet connection... i 
could firewall IPs from my home just like the chinese government can firewall 
sites from all of their citizens.  There is a chance that changing IPs will 
make it work, but theres also a chance the new IP will be firewalled too...

Just google for Great Firewall of China

--
Matt


From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] [EMAIL 
PROTECTED]
Sent: Sunday, May 04, 2008 8:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] China vaults past USA in Internet users - now 
220 million users in China

The statistics you write of maybe true but there is a small downside to this.  
Presently, our website can not be seen in China and we are hosted by 
Godaddy.comhttp://Godaddy.com.  The explanation we receive more than three 
times is China is blocking a number of IP addresses and there is nothing they 
can do!  This is the kicker we work with a Federal Organization here in USA 
who is also hosting at Godaddy.comhttp://Godaddy.com and their website can be 
seen.  The only difference is this ogranization is using a Class A static IP 
address.  Godaddy.comhttp://Godaddy.com can not guarantee if changing to a 
static IP address this situation will change.  One more thing, 
Godaddy.comhttp://Godaddy.com can be seen in China because their website is 
hosted on a separate corporate server.

This is not a gripe but a realty and we are a digium select reseller.  There is 
a consultant in China we work with and our website is translated in Mandarin.  
If anyone has a proposed solution for this we would greatly appreciate the 
dialogue.

Regards,

 Original Message 
Subject: Re: [asterisk-users] China vaults past USA in Internet users -
now 220 million users in China
From: Dean Collins [EMAIL PROTECTED]
Date: Sat, May 03, 2008 11:27 am
To: [EMAIL PROTECTED]mailto:[EMAIL PROTECTED].com

China has surpassed the USA as the No. 1 nation in Internet users.

The number of Chinese on the Internet hit more than 220 million as of February.
http://mobileanalytics.com/forum/index.php?topic=28.0http://mobileanalytics.com/forum/index.php?topic=28.0




I wonder how Americans are going to handle this little turn of events.

What is really interesting in the rest of the article it discusses how the 
percentage of penetration for china is 17% of it's 1.3 Billion population 
versus 71% penetration of the USA's population of 304 million people.

So with China expected to increase another 13 million users this month alone 
(March 2008) to 233 million users how long before there are more people using 
the internet in China than the entire population of the USA (I'm guessing about 
7 months so about the end of this year).

Does anyone in the Asterisk community have a good website for getting accurate 
voip minutes or some other field of reference for how successful voip 
penetration is in the respective countries? Would be interesting to see what 
countries are leading Voip implementation penetration regardless of whether it 
is Asterisk or Avaya etc etc.

I know everyone freaked when Trixbox was collecting stats but I think it would 
be great for someone to write a small ‘anonymous collection module’ that an 
Asterisk sys-admin could download and install on their asterisk server which 
uploaded the stats to a community website.

Even if it just collected number of new installations globally this would be a 
huge help to people selling asterisk to their customers who continually ask 
“I’ve heard about this Asterisk open source stuff but how many are there 
installed globally anyway?”




Regards,

Dean Collins
mailto:[EMAIL PROTECTED][EMAIL PROTECTED]mailto:[EMAIL PROTECTED]
Cognation Limited
+1-212-203-4357
+61-2-9016-4652 (Sydney indial)
P.S. In case you are wondering Australia has a piddling little 15m users but 
this is against a pop of only 20.5m so the Internet penetration is actually 
higher than the 

Re: [asterisk-users] AGI asterisk high balance

2008-05-03 Thread Matt Watson
There is really no reason why you cannot.

Personally... I'd avoid using Java for AGI's that you think are going to 
receive heavy use... simply because the JVM adds a lot of overhead, and 
possibly a very real performance impact from having the load the JVM everytime. 
 Of course there is overhead as well if you do PHP instead, as the PHP 
interpreter has to load everytime... but that's probably pretty light-weight in 
comparison to the JVM.

Of course you could compile your Java code to native binaries to work around 
that problem, but I have no experience doing that.

Please keep in mind that I have not actually created or used any Java AGI's... 
really just my thoughts without any experience.

--
Matt

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chetherston miles
Sent: Saturday, May 03, 2008 6:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] AGI asterisk high balance

Hello,

Is there a problem to use AGI JAVA to write an AGI to billing calls and 
customer accounts?

Anyone have experience with it could give me some tips?

Thanks,

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Re: [asterisk-users] rebooting newer cisco phones

2008-05-02 Thread Matt Watson
I don't know about the Cisco phones...  I;m using Aastra phones which I can 
send a SIP NOTIFY to have them check for updated config... when they detect a 
new config they reboot themselves and download the new config.

But your switch might also have an option to disable PoE on a per-port basis... 
I know our Dell 3548P's do.

--
Matt

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail
Sent: Friday, May 02, 2008 1:26 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] rebooting newer cisco phones

Does anyone have a solution for remotely getting the newer cisco phones (7941, 
7961, 7970, etc ) to reread their configs (or even rebooting).  I am running 
SIP firmware connected to asterisk.

Check-sync doesn't seem to work anymore, I can't login to the phones as root 
because I am given a challenge: random digitspassword: prompt. occasionally 
I need to make changes to phones at remote locations, only solution I have now 
is rebooting the POE switch. Kinda an overkill.
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Re: [asterisk-users] Zaptel 1.4.10.1 Released

2008-05-01 Thread Matt Watson
Does anybody know if this version fixes the soft lockup during ztcfg using a 
TE200B?

http://bugs.digium.com/print_bug_page.php?bug_id=12468


--
Matt


From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Asterisk Development 
Team [EMAIL PROTECTED]
Sent: Thursday, May 01, 2008 1:07 PM
Subject: [asterisk-users] Zaptel 1.4.10.1 Released

The Asterisk.org development team has announced the release of Zaptel
version 1.4.10.1.  This release is a bug fix release for a regression in
which the Zaptel udev rules were not installed correctly, as well as a
few minor fixes in the xpp drivers.

This release is available as a tarball as well as a patch against the
previous release.  It is available for download from downloads.digium.com.

Thank you for your support!

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Re: [asterisk-users] Zaptel 1.4.10.1 Released

2008-05-01 Thread Matt Watson
err, that should of read TE220B




From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Matt Watson [EMAIL 
PROTECTED]
Sent: Thursday, May 01, 2008 3:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zaptel 1.4.10.1 Released

Does anybody know if this version fixes the soft lockup during ztcfg using a 
TE200B?

http://bugs.digium.com/print_bug_page.php?bug_id=12468


--
Matt


From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Asterisk Development 
Team [EMAIL PROTECTED]
Sent: Thursday, May 01, 2008 1:07 PM
Subject: [asterisk-users] Zaptel 1.4.10.1 Released

The Asterisk.org development team has announced the release of Zaptel
version 1.4.10.1.  This release is a bug fix release for a regression in
which the Zaptel udev rules were not installed correctly, as well as a
few minor fixes in the xpp drivers.

This release is available as a tarball as well as a patch against the
previous release.  It is available for download from downloads.digium.com.

Thank you for your support!

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Re: [asterisk-users] Hyperthreading and multicore

2008-04-29 Thread Matt Watson
This is my understanding of hyper threading, which I believe to be accurate.

Basically, as some have mentioned previously, the OS 'sees' your single 
physical core processor as 2 logical processors, in generally, logical 
processors are treated exactly as if they were real processors, and in the case 
of many OS's. they probably don't understand the difference - Linux however 
does have specific SMT support for hyperthreaded cores.

Basically not all CPU instructions take the same amount of clock cycles to 
complete, some may take 3, some may take 7, etc.

Many of these clock cycles actually goto waste because the CPU is waiting for 
something, for example, an instruction that involves a fetch from memory, if 
this takes 7 clock cycles to complete, 4 of those cycles might go wasted while 
the CPU essentially just sits there and waits for the data to be fetched form 
RAM, L1 or L2, or L3 cache.

Hyperthreading essentially puts these wasted CPU cycles to use by allowing the 
CPU to execute a separate thread while it would otherwise be idle waiting.

To me Hyperthreading is an excellent technology... I;m all about efficiency and 
trying to maximize resource usage whenever possible... and that exactly what 
hyper threading does.

That all being said... Hyper threading should not be thought of as effectively 
doubling your CPU power... as previous posters have said, Hyper threading will 
result in single threaded applications actually running slower.. this is 
because you still have other background processes running which may run on the 
other logical processor which could steal CPU cycles away from your main 
application... since you essentially have 2 threads executing on the same 
physical core, there are going to be times when one thread has to wait extra 
clock cycles while the other thread is executing.  Remember its only those 
normally wasted clock cycles that you are going to gain a performance boost 
out of by making use of them... only 1 thread can actually be executing at any 
given time, so the CPU has to schedule these and try balance the threads 
equally so they each get an equal share of the physical core.

I can't say how Asterisk behaves or makes use of additional cores or if hyper 
threading is advantageous to Asterisk or not... I don't know enough about the 
low level parts of Asterisk enough to make an informed opinion about that.

I just thought I'd throw in my 2 cents about what hyper threading is and what 
it does.

--
Matt


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis
Sent: Tuesday, April 29, 2008 9:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hyperthreading and multicore

Matt Florell wrote:
 Also, I have heard HT processors explained this way, on an HT
 processor it's like running 2 virtual processors at 70% of the specs
 of the processor with HT turned off. It's not really like that in all
 situations, but overall it has held pretty much true for me in most
 non-Asterisk situations. Asterisk didn't benefit much from having HT
 enabled on a P4 with HT capability.

That wouldn't surprise me - after all, HyperThreading works on the
principle of allowing two threads to use different dedicated processor
resources (such as floating point math processors and so on) at the same
time... however if two threads are trying to use the same processor
resource, one thread will be suspended until that resource becomes
available.

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Re: [asterisk-users] choopy audio when both side talk at the same time

2008-04-25 Thread Matt Watson
You might want to begin with tuning your rxgain and txgain settings... there 
are a few methods for doing this on the internet, unfortunatly nobody can give 
you exactly values to use for tx/rxgain as they will be not only specific to 
your install, but specific to every single analog line you have... you can 
probably get away with setting it once for all of your lines, but i'd recomend 
setting it for every one.

http://lists.digium.com/pipermail/asterisk-users/2004-November/064312.html

That is the best guide i;ve seen to doing it, but there may be better ones.

After you get rxgain and txgain tuned for every one of your lines, you'll 
probably notice a dramatic decrease in echo right away, but you can also tune 
your echocancel= and echotraining= after that.

You can set these values on a per-channel basis by doing it like:

rxgain=6.3
txgain=-1.0
channel = 9

rxgain=7.595
txgain=-2.0
channel = 10

rxgain=6.3
txgain=-1.136
channel = 11

etc.

First step IMHO is getting your rx/txgain set properly... don't underestimate 
how important those values are... I learned that the hard way.

--
Matt

From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Ruben Zamora [EMAIL 
PROTECTED]
Sent: Friday, April 25, 2008 7:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] choopy audio when both side talk at the same time

Hi

I have a server with the last version of asterisk branches, zaptel
branches, 2 Digium Card with TDM800P
16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10
Grandstream GXP2000.

zapata.conf
echocancel=64
rxgain=0
txgain=0

when i place a call o receive a call, I finish a sentence i hear a
,AND  when the both side talks at
the same time i have choppy audio.

Any help i appreciate.

Thanks Ruben

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Re: [asterisk-users] Digium B410P or Sangoma A502D?

2008-04-24 Thread Matt Watson
I haven;t used any BRI cards but... call me crazy but wouldn;t they still be 
using Zaptel (even your sangoma... the script might just be configuring it for 
you)...

and btw, software echo cancel happens in the zaptel kernel driver... it has 
nothing to do with the hardware (hence why its a software echo cancel)

You also would of had the option of buying HPEC licenses for software echo 
cancel from digium for a rather cheap price.

--
Matt

From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Andres [EMAIL PROTECTED]
Sent: Thursday, April 24, 2008 5:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Digium B410P or Sangoma A502D?

We have tested both and they work fine.  The Sangoma is much easier to
install as it does not depend on any other driver, you just run
'setup-sangoma' and follow the instructions.  You don't have to fiddle
with the linux kernel or  zaptel or chan_misdn.  It just works.  Plus
its more modular.  You can chose 2/4/6 ports to buy and if you need more
just add remoras up to 24 ports.  The Digium card is fixed to 4 ports,
period.

Having said that, make sure you stick with the version that has hardware
echo cancel and not even try the other one.  We made the mistake of
buying the first time without echo cancel expecting to test the
'software echo cancel'.  But there is no such thing as 'software echo
cancel' on this card.  I do not even understand why Sangoma would make a
version without the hardware echo cancel.  You get some degree of echo
on practically every call.

Andres.



Patrick wrote:

Hi,

I need to setup an Asterisk box with 4x ISDN BRI links. Looking at the
specs of various cards I favor the Digium B410P and Sangoma A502D
because of hardware echo cancellation. Does anyone have any experience
with either card, good or bad? Which one would you choose and why?

Thanks for your insight.

Regards,
Patrick


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Re: [asterisk-users] Switch recommendation?

2008-04-22 Thread Matt Watson
I'm using Dell 3548P switches currently which I have powering 25 phone (mostly 
Aastra 9133i's, a couple 480i's a 57i + 560M)

So... basically I have a phone on about half my ports... my power utilization 
on the switch is:

console# show power inline

Unit  Power  Nominal Power   Consumed Power   Usage Threshold   Traps
 --- - -- --- -
 1 On  370 Watts 78 Watts (21%) 95 Disable


The unit will supply up to 370W of power, or 470W if you buy the additional 
power supply for it.


--
Matt

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan C. 
Bailey
Sent: Monday, April 21, 2008 9:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Switch recommendation?

We've been using D-Link DES-3028P and DES-3052P switches. They can supply full 
power to EACH port unlike the Linksys switches we've tried. They're also rock 
solid from our experience.


-Jon

- Original Message -
From: Hilary Miller [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, April 21, 2008 8:21:12 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-users] Switch recommendation?

On Mon, Apr 21, 2008 at 5:54 PM, Sean Dennis [EMAIL PROTECTED] wrote:
  The Cisco 3524 switch doesn't support 802.3af which is what your Linksys
  phones are going to want.

Thank you for sharing Sean! When I saw them I felt a disturbance in
the force, and now I know why!

--
Just Hil

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