Re: [asterisk-users] ​ PJSIP and Non Media Proxy

[Pete Mundy]

> On 6/11/2017, at 7:42 AM, Saint Michael  wrote:
> 
>  I see here a big disconnect between Digium and the VOIP industry. 99% 
> of the VOIP entrepreneurs like me would need to avoid proxying the media. 
> 

Wow, that's quite a bold statement. I must be one of the 1% then because I'm a 
VoIP entrepreneur and I've had no need to do such thing.


>  I mean people like me buy and sale billion of minutes every day" 
> 


Granted, I don't "buy and sell billions of minutes every day"... By my calcs 
that would require some 11K+ channels running 24X7, and that's just for one 
billion per day! So I'm dubious of the claim, and that in turn makes me dubious 
of the quoted 99% figure too.


Pete




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Re: [asterisk-users] ERROR during high volume MoH dialplan

[Pete Mundy]

>> On Thu, 31 Aug 2017, Joseph Smith wrote:
>> 
>> So I am looking for a better way to allow several thousand callers to listen 
>> to this IVR menu at the same time.


> On 1/09/2017, at 7:10 AM, Steve Edwards  wrote:
> 
> I'm thinking multiple hosts.
> 
> I'm not a fan of 4,000 eggs in one basket.


+1 for horizontal scaling as the best solution in this situation.

Pete



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Re: [asterisk-users] OT: Want to capture all SIP messages

[Pete Mundy]

> On 1/06/2017, at 9:24 AM, Jeff LaCoursiere  wrote:
> 
> On 05/31/2017 04:13 PM, Steve Edwards wrote:
>> On Wed, 31 May 2017, Barry Flanagan wrote:
>> 
>>> sngrep
>> 
>> Isn't sngrep a great tool? Since discovering it my use of tcpdump/wireshark 
>> has cratered.
>> 
>> Being able to compare an INVITE that worked with one that didn't (with color 
>> highlighting) rocks.
> 
> On sites where I want an always available packet history I use tcpdump with 
> the -C and -W options to manage a ring buffer of X bytes.  Then you can use 
> cool tools like sngrep or really anything that operates on pcap files at whim.
> 
> Cheers,

Heya Steve

I use the same Jeff recommended.

Eg this command would capture SIP traffic in capture files up to 100Mbytes 
each, with a maximum of 10 files in play and overwriting the oldest 
automatically:

tcpdump -i eth0 -w rollingSIPtrace. -C 100 -W 10 port 5060

Eventually you'd end up with files called 'rollingSIPtrace.00' through to 
'rollingSIPtrace.09', and when rollingSIPtrace.09 reaches 100MB, overwriting of 
rollingSIPtrace.00 (then rollingSIPtrace.01 etc) would commence.

Does that achieve your goal?

Or was the problem that if your server restarts and the command auto-executes 
at boot time then the first file overwritten will be rollingSIPtrace.00, not 
necessarily whichever file was the last modified?

Pete




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Re: [asterisk-users] have you heard the news?

[Pete Mundy]


https://en.wikipedia.org/wiki/On_the_Internet,_nobody_knows_you%27re_a_dog

?


> On 30/04/2017, at 7:55 pm, Marco Signorini  wrote:
> 
> <17D08BA890CCA80D6E89EFEEED63F242.jpg>


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Re: [asterisk-users] log incoming calls without answering

[Pete Mundy]

> On 21/04/2017, at 9:33 am, Fabio Moretti  wrote:
> 
> the point basically is: it is possibile for asterisk to log a call
> without answering it? How to do it in the dialplan? Or I'm wasting time
> because an analog line who enter asterisk is always answered?

Yes.

Something similar to this will work (untested, possibly with syntax errors, but 
gives the idea):

exten => s,1,NoOp(Inbound call from callerID $CALLERID(num))
exten => s,n,Hangup

The 'Hangup' will simply tell the FXS box (which you've used to terminate the 
line) that Asterisk isn't accepting the call.

One problem could be the FXS box re-inviting Asterisk over and over (causing 
duplicate NoOp logs), but you could add a Wait in there for 2-3 minutes before 
'Hangup'. Not many calls would ring for more than 2-3 minutes. In fact the PSTN 
may limit the number of rings (I have a feeling here in NZ it's about 250 rings 
or so).

exten => s,1,NoOp(Inbound call from callerID $CALLERID(num))
exten => s,n,Wait(180)
exten => s,n,Hangup

Pete



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Re: [asterisk-users] log incoming calls without answering

[Pete Mundy]

Fabio, this doesn't answer your question directly and it's not Asterisk related 
in any way, but it's another way to engineer a solution to the problem and I've 
seen it done before.

Many analog modems will decode the caller ID on the analog line and provide it 
as part of the 'RING' notification on their serial port. This can be used as 
input on a script on a computer to log the caller ID and datestamp.

For the Asterisk solution you propose to be able to work, the analog line has 
to support caller ID. If it does that then it will work with any of these 
modems too.

It's a purely passive connection to the line (just bridge it over the pair) and 
it may be cheaper/simpler/easier for you to build.

Food for thought?

Pete



> On 21/04/2017, at 8:26 am, Fabio Moretti  wrote:
> 
> Hi,
> 
> I've some analogic lines and I'm asked if it's possible to program an 
> asterisk for "checking" the inbound calls without answering them, doing 
> something like this:
> 
> analog line 1 -+-- asterisk
>   |
>   \__ analog phone
> 
> when a call enter, asterisk sense it and store its values (callerid, date and 
> time, etc) somewhere, but nothing more, people will answer using the old 
> analog phone.
> The goal is to have a log of the inbound calls without touching the old 
> analog system (it's shared between different subjects).
> 
> I'm pretty sure it's something possible, but how to tell asterisk: "ok, call 
> this AGI, and then don't answer and do nothing more".
> 
> Any idea?
> 
> Thanks


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Re: [asterisk-users] Voicemail asking for login

[Pete Mundy]

> On 19/04/2017, at 4:25 pm, D'Arcy Cain  wrote:
> 
>> Does this mailbox exist?
> 
> Yes.

Hmm... Above my pay grade I'm afraid! Looking at your 'voicemail show users' I 
can't see why the vm_authenticate function is failing to read the username :(

If I were any good at coding in C, I'd probably look inside app_voicemail.c 
around line number 10671 and see if I could determine how it reads the username 
and maybe throw some hacky debug output in there to try and determine at which 
point of that process it's failing. But I'm no good at coding in that language, 
so will have to defer to others to help.

Good work on sending through the console clipping and relevant info. Sorry I 
couldn't resolve it for you.

Anyone else got any other ideas?

Pete




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Re: [asterisk-users] Voicemail asking for login

[Pete Mundy]

> On 19/04/2017, at 7:58 am, D'Arcy Cain  wrote:
> 
> 
> Everything looks the same as another one that works except for two things.  
> The one that works doesn't have the "Probation passed" lines. I am not sure 
> if that is even part of this call.  The other is the line with "Playing 
> 'vm-login.gsm'" in it.  at that point the working one has this:
> 

Presumably also the line containing 'vm_authenticate: Couldn't read username' 
also doesn't appear in the output on a working mailbox either?

I think that's the place to concentrate your efforts.

It shows shortly after the attempt by VoiceMailMain to enter mailbox 
'stocktrans2' in context 'VoiceMail'. Does this mailbox exist?

Can you show the equivalent line from a working mailbox (so we can see if it 
also uses the context 'VoiceMail', or maybe something else instead, like 
'default'?).

Pete



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Re: [asterisk-users] Voicemail asking for login

[Pete Mundy]

Hi D'Arcy

> On 18/04/2017, at 5:17 am, D'Arcy Cain  wrote:
> 
> 
> One user (that we know of so far) has a different experience.  In that case 
> they are asked for a mailbox number first.  
> 
> I have tried searching for this issue but nothing seems to apply.  Most 
> discussions are about "*97" vs. "*98".  Can anyone suggest another field of 
> enquiry?


Try this:

asterisk -r
core set verbose 10
[get user to trigger fault]
[examine console output, and post to list if still unclear]

If you don't solve it yourself, then we'll be able to help further once we've 
seen the output.

HTH,

Pete




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Re: [asterisk-users] Define SIP fromuser field in Dial()-command

[Pete Mundy]

Hi Jonas 

Does the information at this link help? 

http://the-asterisk-book.com/1.6/funktionen-callerid.html

Pete


> On 5/04/2017, at 8:11 pm, Jonas Kellens  wrote:
> 
> Hello
> 
> anyone have some useful input on this ?
> 
> 
> 
> Thanks.
> 
> 
> On 03-04-17 10:25, Jonas Kellens wrote:
>> Hello
>> 
>> how can I set the fromuser field of the SIP INVITE when using the 
>> Dial()-command in the dialplan ?


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Re: [asterisk-users] moh reload not reloading/reading new musiconhold files

[Pete Mundy]

Hi Jonas

Wouldn't this do the job?

touch /etc/asterisk/musiconhold.conf ; asterisk -rx 'module reload 
res_musiconhold.so'

Pete


> On 31/03/2017, at 8:55 am, Jonas Kellens  wrote:
> 
> 
> I would not know how to automate this through script...
> 



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Re: [asterisk-users] First SIP-registering succeeds, second doesn't

[Pete Mundy]

+1! This sounds an awful lot like an ALG doing it best to 'help'...


> On 14/02/2017, at 6:38 am, Israel Gottlieb  wrote:
> 
> Disable all sip alg/helpers in the router



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Re: [asterisk-users] Disallow CALLS without registry

[Pete Mundy]

> On 12/02/2017, at 7:27 am, Dave Platt  wrote:
> 
> Now, there probably _is_ a way to force specific peers to register
> prior to placing a call, if that's what you really want to do (although
> I would ask "Why?" to anyone who wants to do things this way).  The
> way I would do it, in Asterisk, is:`
> 
> 

Good thinking Dave, and thanks for adding your comments to the discussion. I 
love how Asterisk is like it's own programming language; if the fundamental 
components your feature needs are available from the system but the glue that 
provides the logic doesn't exist, then you can just write it yourself!

Excellent suggestion :)

Pete



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Re: [asterisk-users] Disallow CALLS without registry

[Pete Mundy]

> On 11/02/2017, at 3:40 am, Frank Vanoni  wrote:
> 
> On Thu, 2017-02-09 at 14:58 +0200, Антон Сацкий wrote:
> 
> 
>> so the main question is -- how to Disallow CALLS without registering
>> on PBX
> 
> sip.conf configuration
> In the [general] section, define:
> 
> 
> [general]
> ...
> allowguest=no
> alwaysauthreject=yes
> ...

I don't think either of these configuration attributes achieve what the OP 
requested ("how to Disallow CALLS without registering on PBX").

In fact, I'm not sure that it's actually possible to disallow [authenticated] 
calls from a peer that hasn't registered!

As far as I can tell, 'registration' was never intended to be part of the 
authentication process. It's sole purpose is to inform the PBX as to the 
current location of the endpoint. I suspect this means that what the OP is 
asking for cannot be achieved with the current code bases.

But each time I'm proven wrong I learn something, so if I'm wrong then please 
by all means correct me! :)

Pete




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Re: [asterisk-users] Call List Campaign to an IVR

[Pete Mundy]

On 2/02/2017, at 9:52 pm, A J Stiles  wrote:
> 
> but in simple solidarity with everyone who has ever 
> been pissed off by a machine-initiated spam marketing phone call at an 
> inappropriate moment, I am not going to tell you how to do it.
> 

Hat-tip to you, AJ :)

Pete



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Re: [asterisk-users] GotoIf Double Pattern Match [SEC=UNCLASSIFIED]

[Pete Mundy]

> On 28/11/2016, at 12:29 pm, Calum Power  wrote:
> 
> 
> 
> What I have written come up with is below, but I just wanted to see if I was 
> going about this the right way, and that this expression would actually 
> work...
>  
> exten => B,n,GotoIf($["${CALLERID(num):-2}" = "13"] & 
> $["${CALLERID(num):0:2}" = [5-9][5-9])?change_context)
>  
> Any advice/suggestions/flames welcome J

Hi Calum

I think you'll need another closing square-bracket at the end of the second 
conditional test in your example, ie:  5-9]])?cha

I also think you may need to wrap that second conditional up in the REGEX 
function, although there are much smarter people on-list who will likely come 
back with a better suggestion!

Also, checking the second character of CID is the same as the first character 
(otherwise you'd match 56, 69, 97 etc).

I'm thinking something like:

$[${REGEX("[5-9]" ${CALLERID(num):0:1})}] & $[${CALLERID(num):0:1} = 
${CALLERID(num):1:1}]

Hope this helps anyway :)

Pete







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Re: [asterisk-users] Touch tone stutter

[Pete Mundy]

One direction that may be worth exploring further is his ATA's config (or 
perhaps swapping it for a different model). Eg adjusting echo cancellation or 
line impedance settings.

Is the ATA he is using the same as the ATA you use? 

Failure to correctly recognise and decode DTMF is just one of many reasons why 
I never use them (ATAs). Like faxing over VoIP, they're just too much trouble 
:( 

Genuine IP phones are pretty good value these days. Could you drop one of those 
on-site as a temporary measure to prove that it's phone and/or ATA related?

Pete

Ps, you might also want to consider joining VoiceOps (if you're not already 
subscribed) and posting there. https://puck.nether.net/mailman/listinfo/voiceops


> On 23/11/2016, at 12:16 pm, D'Arcy Cain  wrote:
> 
> I am hoping someone else has seen this and can offer a solution or at least a 
> direction to investigate.  I am running 11.23.  Most of my clients are fine 
> but one has a strange behaviour.  He has a Grandstream HT701 like most of my 
> clients who use an ATA.  He can make call and they are crystal clear.  
> However, when he tries to use phone menus ("dial 234 for John Doe" for 
> example) it doesn't work.  At first I thought that the tones were not being 
> delivered but I had him play them to me and the issue is that each tone 
> stutters.  As a result, entering "234" becomes "223344" which is not 
> understood as a valid input.
> 
> He has a recent phone and, in fact, is almost the same model I have at home.  
> His is a Panasonix TX-TGD220 and mine is a TX-TGD-212.  The difference is 
> mainly that his has a built in answering machine.
> 
> As I said, no one else is having the issue.  One person has a horrible 
> connection with voice drops all the time but the touch tones still work.
> 
> I have made two files available.  http://darcy.vex.net/Bishop.ogg  is an OGG 
> file of the sound that it makes at the receiving end and the other at 
> http://darcy.vex.net/Bishop.png is a picture of the wave form.  I had the 
> user think "one Mississippi" etc. and alternately press and release three 
> different buttons.  I recorded off my SIP phone which is going through the 
> same Asterisk server as the user.
> 
> The only thing I can see in my configs that might apply is in sip.conf 
> "dtmfmode=rfc2833" which I don't want to change unless I am absolutely sure.  
> No one else is having the problem so I don't want to risk it. Would I be safe 
> if I set it to "auto"?  Is there any chance that it would help?  Is there 
> some place else I should be looking?
> 
> Thanks in advance for any help.
> 
> -- 
> D'Arcy J.M. Cain
> System Administrator, Vex.Net
> http://www.Vex.Net/ IM:da...@vex.net
> VoIP: sip:da...@vex.net


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Re: [asterisk-users] RS485 Audio device

[Pete Mundy]

Unless there is already RS485 in place, forcing the use of that type of bus, 
this sounds to me like something that would be more easily achieved using one 
of those 2-wire SIP doorphones that puts standard analog audio over a copper 
pair between the handset and the base.

I don't have any specific model to reference to, but they've definitely been 
discussed on-list before.

Pete


> On 3/11/2016, at 8:46 am, Jerry Geis  wrote:
> 
> Hi All,
> 
> The reason for the question was simply that the customer desired some solution
> called an "AOR" or Area of refuge - I think it was. Basically a call button, 
> microphone and speaker to hear back
> with the kicker being "a long distance" the solution has to run.  RS485 is 
> like 4000 feet.
> 
> There are solutions our there apparently that are not built on asterisk - so 
> I was just trying to find
> a solution that potentially worked with asterisk. 
> 
> Thanks! 
> 
> Jerry


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Re: [asterisk-users] RS485 Audio device

[Pete Mundy]

It caught my interest for the same reason! It's such an obscure query.

My guess was that the desire was to run it over an existing shared RS485 bus, 
which means the maximum data rate available would be even less because it would 
be shared with other devices.

The only way I could imagine a solution working in this scenario would be some 
sort of ATA attached to a router which supported PPP and had an RS232 serial 
port on the WAN side (eg like an old modem router), which could then be 
converted into RS485. You could possibly do all that in a Raspberry Pi with a 
USB-RS232 adapter (or even a USB-RS485 adapter directly!). Then you'd run SIP 
over IP over PPP to a similar device on the other end of the bus. But you'd 
obviously have to be sure to use a low bandwidth codec, and you'd possibly 
suffer audio quality issues when there was (RS485) bus contention.

But that was all guess work. It would be great if OP Jerry could expand a 
little more on the application scenario, even if just to whet the curious 
appetites :)

Pete

> On 3/11/2016, at 3:48 am, Telium Technical Support  wrote:
> 
> This one caught my interest too...more out of curiosity!  Keep in mind that
> RS485 max speed drops to 100kbps after a relatively short distance.  And,
> 100kbps is RAW speed.  If you encapsulated your audio stream in that you'd
> lose another 10%.
> 
> So why are you doing this?  If you are running a 100m cable (4 wire +
> shield) why not just pull at cat5/6 cable instead?  Or just send analog
> audio over 2 of the wires with the shield to keep out hum.   If there is a
> need to use rs485 you could stream your audio over that connection - but I'm
> curious why first.
> 
> We did some work for broadcast (radio station) doing AoIP and converting
> some analog feeds but this seems unusual.
> 
> Jason



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Re: [asterisk-users] Multiple readfile oddities, newlines etc

[Pete Mundy]

On 18/10/2016, at 10:38 am, Steve Edwards  wrote:

>> cat /home/test/feature-1.txt | hexdump
> 
> Or just:
> 
>   hexdump /home/test/feature-1.txt

Heh.. yes, fair call ;)

Pete



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Re: [asterisk-users] Multiple readfile oddities, newlines etc

[Pete Mundy]

> On 18/10/2016, at 2:31 am, Jonathan H  wrote:
> 
> I have a plain text file, ASCII, unix line breaks. 1 single line, and all 
> that is in it is the word "radio".
> 
> 

Heya Jonathan

Interesting problem!

Unfortunately I can't help with suitable dialplan code to resolve your issue, 
but I thought I'd at least throw this snippet out to you:

If you want to know what is _really_ in that file (including all invisible 
characters and anything else that wc etc might not count), pipe it through 
'hexdump'. Then you get to see the raw contents of the file in hex and can 
compare it to other files or the output you get from other tools (and you can 
look up the ASCII char codes for invisible chars to explore what they are etc).

Eg:

cat /home/test/feature-1.txt | hexdump

Hope this tool helps you in your quest :)

Pete





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Re: [asterisk-users] SIP on multiple ports

[Pete Mundy]

> On 18/10/2016, at 12:30 am, Jerry Geis  wrote:
> 
> I am running iptables on the 10.201 machine. I have not control over the 
> other machine. It is a microsoft lync product.
> 
> my definition...
> [MyTrunk]
> type=friend
> dtmfmode=rfc2833
> disallow=all
> allow=ulaw
> allow=alaw
> context=my-incoming
> host=192.168.1.3
> ;port=5068
> canreinvite=yes
> qualify=yes
> transport=tcp
> 
> I have tried it with or without the port=5068.



Heya Jerry

With port=5068 and transport=tcp both defined, do calls outbound to the Lync 
device fail, or is it just registrations FROM it that are failing?

I seem to recall you mentioning it's registrations that aren't working. However 
for those to work you would need your Asterisk on 10.201 to also be listening 
for registrations via TCP, and on port 5068. I'm thinking this would be a 
global configuration definition, not a peer-specific definition. Others may be 
able to chime in with the specific options required...

Pete

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Re: [asterisk-users] iptables for SIP talk to other port

[Pete Mundy]

Jerry has already clarified in a previous reply that he is running SIP over 
TCP, not UDP.

But he hasn't clarified on which machine he is applying the iptables header 
rewrite rules (10.201, or 1.3?).

Either way though, it seems like a kludgy work-around. IMO, it'd be better to 
focus on creating the correct Asterisk peer configuration for the peer that is 
operating on the non-standard separate port, and don't use any packet-header 
mangling at all.

Jerry, can you post your configuration for the peer in Asterisk? (eg from 
sip.conf)

Pete


> On 17/10/2016, at 12:27 pm, Duncan  wrote:
> 
> 
> Don't you want udp rather than tcp?
> 
> Have a look at the iptables stats to see if any packets are hitting your 
> rule. 
> Also I think the source port from your host will be 5068 so your replies will 
> be to the right port but you can double check
> 
> tcpdump is also very useful here
> 
> sudo tcpdump -i eth0 -n udp and host 192.168.1.3 should show you packets 
> between your machine and your odd host
> Cheers Duncan
> 
> On 17/10/16 11:55, Mike wrote:
>> 
>> I'm by no means an iptables guru... 
>> 
>> Not sure if it's necessary to enable forwarding via: 
>> echo "1" > /proc/sys/net/ipv4/ip_forward 
>> 
>> Also have you tried without the "POSTROUTING" rule? 
>> 
>> I seem to recall that "iptables" is smart enough to correctly route packets 
>> back out without that rule. 
>> 
>> 
>> On Sat, 15 Oct 2016, Jerry Geis wrote: 
>> 
>>> I have a host 192.168.1.3 that wants to run SIP on 5068 (long story).My 
>>> host is 192.168.10.201. 
>>> My host needs to stay on 5060 because of all the other devices I have 
>>> connected. 
>>> 
>>> I tried putting port=5068 in my SIP extension definition but that did not 
>>> work. 
>>> 
>>> So I thought about using iptables to accomplish this: 
>>> 
>>> iptables -t nat -A PREROUTING  -p tcp --dport 5068-j 
>>> REDIRECT --to-port 5060 
>>> iptables -t nat -A POSTROUTING -p tcp --dport 5060 -d 192.168.1.3 -j 
>>> REDIRECT --to-port 5068 
>>> 
>>> 
>>> Do I not have the right format of the command? 
>>> Anything incoming destined for 5068 redirect to 5060... 
>>> Anything going out to 192.168.1.3 and port 5060 redirect to 5068. 
>>> 
>>> Seems like that should have worked? 
>>> 
>>> Thoughts?  sip show peers still says unreachable. 
>>> 
>>> Thanks, 
>>> 
>>> Jerry 


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Re: [asterisk-users] asterisk admin interface

[Pete Mundy]

> On 17/05/2016, at 9:55 am, Goke Aruna  wrote:
> On May 16, 2016 22:15, "Telium Technical Support"  > wrote:
> > 
> >
> > In this case a very simple solution is to modify the Asterisk config files
> > to add/remove users, then tell Asterisk to reload from the CLI/AMI.  And
> > that's it!
> 
> Thanks Raj
> 
> You are correct. Is there any open source application in that?
> 


According to WikiPedia, there are open-source implementations of vi available:

https://en.wikipedia.org/wiki/Vi


Pete




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Re: [asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements

[Pete Mundy]

On 29/04/2016, at 3:46 am, A J Stiles  wrote:
> 
> 

> 
> There is no reliable way to distinguish whether a phone was answered by a 
> human being or a machine.  
>  
> If you can't just disable voicemail on all your SIMs then you will need to 
> find out how long each carrier will wait before diverting to voicemail, and 
> then make sure the timeouts in your Dial() statements are short enough not to 
> trigger the carrier's voicemail.  Then use Asterisk's VoiceMail() application 
> to record any message your caller might leave.
> 
> 


To make things worse, in the country I reside not one of the cellular telcos 
have the option of disabling voicemail! If you disable voicemail they still 
terminate the call and instead play a message to the effect of 'This user has 
disabled voicemail. Goodbye'.

We also can't rely on timeouts because if the destination cellphone is out of 
coverage or turned off then the telco will drop to VM (or no-VM warning) almost 
immediately.

Unfortunately for us, the only option is to code up a routine that checks for 
acceptance confirmation from the destination human, a-la what John was 
describing in FreePBX.

Pete



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Re: [asterisk-users] AMI: check if the user has a Mailbox

[Pete Mundy]

Hi Luca

Would greping for the existence of the mailbox number in /etc/voicemail.conf do 
the trick?

Pete


> On 22/04/2016, at 7:34 am, Luca Bertoncello  wrote:
> 
> Hi list!
> 
> On an Asterisk-Server I have some users. Just two of them have a Mailbox.
> I want to write a little Web interface to manage many things and I'd like to
> have a menu point for the voicemail, but just if the user has a Mailbox.
> 
> I found the AMI-Command MailboxStatus, but it does not return what I need,
> since it returns 0 if the user has a Mailbox but no messages and if the user
> has no Mailbox...
> 
> Could someone suggest me a way to get this information?
> 
> Thanks a lot!
> Luca Bertoncello
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Re: [asterisk-users] Recommendations for free virtual server tech and Asterisk?

[Pete Mundy]

List,

Might as well throw my hat in the ring!

I can't say it's the 'best' way to do it, but I've been running Asterisk VMs 
inside the free 'VirtualBox' software for many years with nill issues (well, 
nill related to the hypervisor environment itself anyway!).

https://www.virtualbox.org

Pete

> On 8/04/2016, at 2:09 pm, Carlos Rojas  wrote:
> 
> I have tried with xen and kvm both are working fine.
> 
> On Wed, Apr 6, 2016 at 3:44 PM, Loic Chabert  > wrote:
> Hello,
> 
> Work well with kvm and centos 7.
> Some ajustements has to be made with systemd.
> 
> I'm using it in production since 1.5 year now, no issue to report.
> 
> Regards.
> 
> Le 6 avr. 2016 21:13, "Yves" > a 
> écrit :
> 
> Le 06/04/2016 18:12, Markos Vakondios a écrit :
> 
> Good evening,
> 
> My English is limited but if I can help.
> 
> We install Asterisk Version 13.1 on VmWare with Debian 8.2, no
> problem since June 2015, currently I have tested on Unbutu 14.04 but problem 
> with network-manager (problem of stability with Asterisk 1.8.32 and 
> difficulty with routing network-manager).
> 
> I also installed Asterisk on KVM (Debian 8.2) no problem (but not test with 
> dahdi) without particular problem.
> 
> here is my little opinion
> 
> Hello everyone
>> Proxmox and KVM on Ubuntu 
>> 
>> On Wednesday, 6 April 2016, Ryan, Travis < 
>> ry...@oscarwinski.com 
>> > wrote:
>> What is the best virtual server tech (and most stable, etc) to use for a 
>> asterisk virtual hosting environment?
>> 
>>  
>> I have a client that wants to do virtual hosting of Asterisk (only SIP or 
>> IAX, no PRI, etc) and I’m wondering if Xen or something else would be best? 
>> We’d like to stay away from the costs of VMWare if possible.
>> 
>>  
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Re: [asterisk-users] Lost outgoing SIP packets

[Pete Mundy]

Roel,

Just another thought bouncing around... Your ifconfig output was specific to 
eth1. Is there an eth0 too? Is there a chance packets are heading to that other 
interface when they shouldn't be? Running a second tcpdump on eth0 at the same 
time should at least disprove the theory quickly.

Pete


> On 1/04/2016, at 2:59 am, Roel van Meer  wrote:
> 
> 
> Thanks for the heads up, and thanks for thinking with me everyone!

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Re: [asterisk-users] hijacked thread

[Pete Mundy]

Sorry George. You're quite right, that was bad etiquette. I should have started 
a new thread with my reply to the hijack.

Pete

> On 22/03/2016, at 4:04 pm, George Joseph  wrote:
> 
> Now do you mind if we get back to the original purpose of this thread before 
> it was hijacked?
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Re: [asterisk-users] Loss of devices registration (pjsip)

[Pete Mundy]

Good result! Glad it worked for you :)

Pete


> On 22/03/2016, at 9:34 am, somsad khan <ctrlz.netw...@gmail.com> wrote:
> 
> I have added CID name prefix on inbound route. and it works fine :) now I can 
> simply forward five incoming routes to one extension. and as far as I guess, 
> if I add CID name prefix for every number. it should work :) thanks alot  :) 
> 
> On Tue, Mar 22, 2016 at 2:28 AM, somsad khan <ctrlz.netw...@gmail.com 
> <mailto:ctrlz.netw...@gmail.com>> wrote:
> hello Pete Mundy,
> 
> thanks alot for your idea and reply. but unfortunately none of our SIP phone 
> have the facilities to use multiple line and UI.
> 
> I can see incoming numbers on my softphone(Zoiper) when a incoming call hits. 
> I liked your incoming caller ID customize idea.
> 
> Is it possible to add company name with incoming numbers. so that company 
> name or any signal will appear with incoming call numbers, will be easy to 
> identify by employee that call is coming into which number. 
> 
> thank you   
> 
> On Tue, Mar 22, 2016 at 2:12 AM, Pete Mundy <p...@fiberphone.co.nz 
> <mailto:p...@fiberphone.co.nz>> wrote:
> 
> Many desk phones support multiple simultaneous SIP registrations. You could 
> use BLF buttons for each SIP registration and the operator uses the LEDs as 
> their queue as to which account is ringing. Alternatively the phone's UI may 
> be able to indicate which account is ringing without the need for BLFs.
> 
> Another option is to re-write the CALLERID(num) or CALLERID(name) to indicate 
> the inbound line (eg prepend a string or number).
> 
> Hopefully that gives you some food for thought :)
> 
> Pete
> 
> 
>> On 22/03/2016, at 8:49 am, somsad khan <ctrlz.netw...@gmail.com 
>> <mailto:ctrlz.netw...@gmail.com>> wrote:
>> 
>> I have a client coming who wants to assign 5 different numbers to one 
>> virtual employee SIP phone at his desk or softphone (Zoiper).
>> 
>> 
>> please let me know if there is any possible ways. 
>> 
> 
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Re: [asterisk-users] Loss of devices registration (pjsip)

[Pete Mundy]

Somsad,

Yep. That's why I suggested it as another option :)

These links may help:

http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
(see CALLERID(num) and CALLERID(name))

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

Pete

> On 22/03/2016, at 9:28 am, somsad khan  wrote:
> 
> Is it possible to add company name with incoming numbers. so that company 
> name or any signal will appear with incoming call numbers, will be easy to 
> identify by employee that call is coming into which number. 


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Re: [asterisk-users] Loss of devices registration (pjsip)

[Pete Mundy]

Many desk phones support multiple simultaneous SIP registrations. You could use 
BLF buttons for each SIP registration and the operator uses the LEDs as their 
queue as to which account is ringing. Alternatively the phone's UI may be able 
to indicate which account is ringing without the need for BLFs.

Another option is to re-write the CALLERID(num) or CALLERID(name) to indicate 
the inbound line (eg prepend a string or number).

Hopefully that gives you some food for thought :)

Pete


> On 22/03/2016, at 8:49 am, somsad khan  wrote:
> 
> I have a client coming who wants to assign 5 different numbers to one virtual 
> employee SIP phone at his desk or softphone (Zoiper).
> 
> 
> please let me know if there is any possible ways. 
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Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN

[Pete Mundy]

>>  check the system and make sure there really is no firewall like I said

> You were right.

Stick around on the list long enough and you'll realise the truth... he always 
is ;-)

Pete


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Re: [asterisk-users] Rasterisk freeze on 4G link

[Pete Mundy]

These are not Asterisk related questions. It is a common problem. Google is 
your friend. Try something like 'console stalls with big packets'.

To answer your question "why", it's simple. "Because the big packets are being 
dropped".

Pete


> On 4/03/2016, at 7:15 am, Olivier  wrote:
> 
> Yes  "cat /var/log/syslog" also broke my console.
> 
> Why would my console break because of inadequate MTU and other PC on the same 
> location, seem unaffected ?
> Because, they most probably mostly use SMTP and HTTP ?
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Re: [asterisk-users] Rasterisk freeze on 4G link

[Pete Mundy]

Oliver,

Not correct!

Duncan and Toufic are spot-on with their answers.

Pete

> On 4/03/2016, at 5:40 am, Toufic Gmail  wrote:
> 
>> 
>> PS: I was about to determine best MTU value but I always thought a 
>> punishment for a bad MTU value would be a lower throughput, not a screen 
>> freeze. Is it correct ?

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Re: [asterisk-users] No sound with internal calls depending on which phones

[Pete Mundy]

Hi Denis

That advice is correct for disabling RTP support in the phone and if you have 
achieved this then your quoted error about SRTP in the Asterisk console (when 
the call is failing) should no longer be appearing.

This will help show that it was a red herring all along.

The next step (IMO) is to use the Snom's built-in packet capture capabilities 
to grab a packet capture of a failed conversation from each phone then post it 
somewhere with a link to the list so that others can inspect the SIP signalling 
to discover where the issue lies.

You may also need to provide some information about your network configuration, 
IP ranges, firewall etc (a little diagram goes a long way).

For information on how to use the packet capture capabilities on the phone 
refer the Snom user's guide. I'm pretty sure it's well documented.

Hope this helps and look forward to investigating the packet captures for you :)

Pete


On 13/11/2015, at 5:46 AM, (lists) Denis BUCHER  wrote:

> Dear Sam, dear jg, dear Mitul, dear all,
> 
> Thanks a lot for your advices! I had the same idea, but it still doesn't work!
> 
> Maybe I changed the wrong option on the GUI configuration ?
> I went to menu "Setup" > "Identity 1" > "RTP" > "RTP Encryption:" > "off" on 
> both phones.
> And in the configuration I see "user_srtp1!: off"
> 
> Is this right ?



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Re: [asterisk-users] Asterisk encrypted authentication for clients

[Pete Mundy]

Motty,

Isn't this why digest authentication (ie the nonce[1]) is part of the standard 
SIP auth handshake?

Ie, why do you think the password is not already encrypted?

Pete

[1] 
https://andrewjprokop.wordpress.com/2015/01/27/understanding-sip-authentication/
(paragraph starting 'Take a look at the Proxy-Authenticate header and you will 
see a Nonce parameter')


On 30/10/2015, at 10:01 AM, Motty  wrote:

> Thanks Jeff, 
> I don't want SIP over TLS. I would like to encrypt password only, I suppose 
> over TLS.
> 
> On 10/29/2015 01:11 PM, Jeff LaCoursiere wrote:
>> You want SIP over TLS.  That encrypts the signalling.  SRTP and ZRTP encrypt 
>> the actual voice traffic.

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Re: [asterisk-users] Asterisk encrypted authentication for clients

[Pete Mundy]

Hi Motty,

Isn't the whole point of the nonce in a SIP registration to ensure the secret 
doesn't go on the wire in plain-text? Is this not enough, or are you looking to 
hide the username too?

(if so, fair 'nuf, just wondering why :)

Pete

Ps, if so then I think TLS is the missing part of your equation.

On 29/10/2015, at 11:54 AM, Motty  wrote:

> Hello,
> I am searching for a solution to encrypt authentication from Asterisk server 
> to clients. Searching srtp seem to encrypt traffic, I just want client 
> authentication with encryption. Can someone point to the right direction? has 
> anybody used ZRTP? experience with ZRTP?
> 
> Thanks,
> _motty
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Re: [asterisk-users] Reverse one way paging or silent monitoring

[Pete Mundy]

Sam, you need to provide _way_ more information if you want a helpful answer. 
You haven't even told us what type of phone equipment you're using!

Given that this type of feature will be specific to the model in use (and not 
necessarily to Asterisk), that type of information is of primary importance for 
getting a useful answer.

In a general sense I would recommend you look at whether the equipment you are 
using can be commanded via specialised header to set speaker volume to zero. 

Alternatively create a conference room that both endpoints get dropped into, 
but only the monitored end has 'speaker' privileges.

Or just press mute from the callers end (and don't forget the disable the guard 
tone at the remote end; again dependant on your equipment).

Pete


Regards,

Pete Mundy


On 27/10/2015, at 8:41 PM, Sam Basan <sba...@bluebe.net> wrote:

> Hello,
>  
> Paging a phone set the phone to auto answer and open the speaker.
> How can I set the phone to turn on just his microphone and the camera, if 
> available, for security remote controlling?
>  
>  
> Thanks,
> Sam
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Re: [asterisk-users] DTMF talkoff beep (still)

[Pete Mundy]

On 9/10/2015, at 5:16 AM, Jamie Rees  wrote:

> 
> 
> I understand this is DTMF talkoff
> 
> 
> My question is how do people running SIP phone systems mitigate against this?


My personal answer to this question has been to completely avoid the use of any 
ATAs at all. Since taking that approach I have never had to deal with the 
problem again.

Not sure how much practical use that is to you in your own situation, but it 
worked for me!

Pete

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Re: [asterisk-users] Change Asterisk MulticastRTP codec

[Pete Mundy]

Hi Matt

Interesting problem! I'm hoping those with knowledge about the internal 
workings of the Page app and multicast will chime in, although it might pay to 
quote your version of Asterisk).

I don't know enough to answer the question itself, but if it were me I would be 
inclined to just work around it by doing something like piping mp3player 
through sox before sending the data on to asterisk.

I may be able to help you achieve that, so if that's good enough then please 
post more of the multicast page config from your extensions.conf.

Pete


On 1/10/2015, at 6:51 AM, Matthew Murphy  wrote:

> Greetings everyone,
> 
> I was wondering if there was a way to change the codec that Asterisk uses 
> when streaming via MulticastRTP. Or perhaps a way to transcode the multicast 
> stream.
> [SNIP]
> Is there any way to get the MP3Player stream to transcode (as it does on the 
> UNICAST stream) when I try to MULTICAST?
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[asterisk-users] Fwd: ferie estive

[Pete Mundy]

Any chance the list admins could unsubscribe Mr Anzaldi until he gets his 
broken auto-responder fixed?



Begin forwarded message:

 From: davide.anza...@netecom.it
 Subject: ferie estive
 Date: 25 August 2015 2:33:00 PM NZST
 To: Pete Mundy p...@fiberphone.co.nz
 
 Sono assente per ferie e rientro lunedì 31 Agosto.
 Potrò rispondere alle Vostre email al ritorno.
 
 Per comunicazioni urgenti Vi prego di scrivere a supp...@netecom.it
 
 Un cordiale saluto

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Re: [asterisk-users] Does the asterisk support for sending image ?

[Pete Mundy]

On 25/08/2015, at 1:17 PM, Thyda ENG ength...@gmail.com wrote:

 Does the asterisk support for sending image ? if it supports how to config it 
 ?

Hi Thyda. Perhaps you missed Joshua's reply to your question back on 12th 
August?

He said:

What do you mean by image?

Thus-far you haven't answered him. May I recommend that you do?

Pete



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Re: [asterisk-users] How to send Image over asterisk sip

[Pete Mundy]

Thyda,

The term 'image' can be quite ambiguous in computing. For example you could be 
referring to a firmware image for a phone or you could be referring to some 
form of live video channel support. Or something else.

Can you be more explicit as to exactly what you mean by 'image file' and/or 
what it is that you aim to achieve or what you want to see happen?

Pete


On 25/08/2015, at 3:47 PM, Thyda ENG ength...@gmail.com wrote:

 I mean by sending image by using sip channel just like we can send text 
 message and what about sending image file ?
 
 On Wed, Aug 12, 2015 at 6:37 PM, Joshua Colp jc...@digium.com wrote:
 On Sat, Aug 8, 2015, at 07:41 AM, Thyda ENG wrote:
  Dear Sir,
 
 Kia ora,
 
 
  I current have done successfully with sip message over asterisk server ,
  and additionally now I want to send the image between sip using asterisk.
  Could any one share me how to config the asterisk for sending image from
  sip?
 
 What do you mean by image?
 
 --
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com  www.asterisk.org
 
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Re: [asterisk-users] How to send Image over asterisk sip

[Pete Mundy]

Hmm, most phones I've used wouldn't have the capability of displaying a bitmap 
image due to only having minimal monochrome displays.

What sort of end device do you perceive to display these images? Can you give 
links to any devices with support for such things?

I'm assuming you mean only a still-photo image, not video image. Perhaps you 
could use a video channel for this and simply display only a still image 
instead? I do believe that Asterisk has video support, although I haven't 
personally used it.

Hope this helps.

Pete


On 25/08/2015, at 4:11 PM, Thyda ENG ength...@gmail.com wrote:

 I mean by sending the .jpg, or .png or . file.
 
 On Tue, Aug 25, 2015 at 11:10 AM, Thyda ENG ength...@gmail.com wrote:
 Yes, I mean sending image file.
 
 On Tue, Aug 25, 2015 at 10:56 AM, Pete Mundy p...@fiberphone.co.nz wrote:
 Thyda,
 
 The term 'image' can be quite ambiguous in computing. For example you could 
 be referring to a firmware image for a phone or you could be referring to 
 some form of live video channel support. Or something else.
 
 Can you be more explicit as to exactly what you mean by 'image file' and/or 
 what it is that you aim to achieve or what you want to see happen?
 
 Pete
 
 
 On 25/08/2015, at 3:47 PM, Thyda ENG ength...@gmail.com wrote:
 
 I mean by sending image by using sip channel just like we can send text 
 message and what about sending image file ?
 
 On Wed, Aug 12, 2015 at 6:37 PM, Joshua Colp jc...@digium.com wrote:
 On Sat, Aug 8, 2015, at 07:41 AM, Thyda ENG wrote:
  Dear Sir,
 
 Kia ora,
 
 
  I current have done successfully with sip message over asterisk server ,
  and additionally now I want to send the image between sip using asterisk.
  Could any one share me how to config the asterisk for sending image from
  sip?
 
 What do you mean by image?
 
 --
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com  www.asterisk.org
 
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Re: [asterisk-users] asterisk server stress test

[Pete Mundy]

Ah cr@p, sorry Steve, didn't mean to top-post there.


 On 20/08/2015, at 5:23 AM, Markus Weiler markus_wei...@mailworks.org wrote:
 We started the 500 calls and used milliwatt app on the first and record on 
 the second host to check the quality. Alternatively just start 500+ calls 
 and call yourself on top. So you can get a good idea how the quality is.

Markus

That's a fascinating concept!

Can you share any more about how you appraised the data and determined your 
results?

ie once you had the recordings on the second host what did you do do 
computationally score them? Do you look at the decoded (1khz?) waveform or do 
you appraise in another way?

Pete





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Re: [asterisk-users] asterisk server stress test

[Pete Mundy]

Markus

That's a fascinating concept!

Can you share any more about how you appraised the data and determined your 
results?

ie once you had the recordings on the second host what did you do do 
computationally score them? Do you look at the decoded (1khz?) waveform or do 
you appraise in another way?

Pete


On 20/08/2015, at 5:23 AM, Markus Weiler markus_wei...@mailworks.org wrote:

 Am 19.08.2015 um 19:07 schrieb Steve Edwards:
 Please don't top post.
 
 On Wed, 19 Aug 2015, James Cass wrote:
 
 Steve, would you be willing to share that quick bash script?
 
 There's no magic in the script, but here it is, embarrassing myself:
 
cp sample-call-file /tmp/
chmod +x /tmp/sample-call-file
for I in $(seq 1 $1)
do
sudo -u asterisk\
cp /tmp/sample-call-file\
/var/spool/asterisk/outgoing/${RANDOM}
done
sleep 10
 
 Here's what's wrong with this snippet:
 
 1) I don't know why I chmod the 'template.' No idea whatsoever. Alcohol may 
 have been involved.
 
 2) I hate single character variable names. I love alcohol.
 
 3) cp is ill advised. For a testing script, it was easy. For a production 
 application, use mv.
 
 In use, I would execute it specifying how many call files to create, like 
 50. Then, take a look at top, iftop, and vmstat. Lather, rinse, repeat to 
 get to your goal.
 
 
 We started the 500 calls and used milliwatt app on the first and record on 
 the second host to check the quality. Alternatively just start 500+ calls and 
 call yourself on top. So you can get a good idea how the quality is.
 
 Call-Files are explained on 
 http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
 
 Markus
 
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Re: [asterisk-users] PTT push to talk solution

[Pete Mundy]

Hi Jerry

As others have eluded to, the 'PTT' feature can mean different things to 
different people depending on their background.

Is it fair to say that you're looking for a one-touch button which initiates a 
call to the other end and causes the other end to automatically answer in 
speakerphone mode?

If that would foot the bill then have a look at the auto-answer feature of 
Yealink and Snom phones (and others I'm sure). It can be easily triggered by 
adding a SIP header line into your dialplan so that the necessary header is 
included with the invite (this triggers the auto-answer at the remote end).

If you have either of these brands to play with and need the dialplan code just 
sing out.

Pete Mundy


On 7/08/2015, at 3:09 AM, Jerry Geis ge...@pagestation.com wrote:

 I am looking for a push to talk solution does anyone know of a good 
 PTT phone one that works with asterisk.
 
 I'm not talking about polycom fake PPT... I'm talking about a real call
 into Asterisk and having to push a button on a headset or the phone to 
 actually talk. not multicast talk like polycom. 
 
 I wish polycom had a real PTT headset but I cannot find one, I like their 
 phones.
 
 Cisco has a PTT headset but seems only for 7960 model. Those phones are
 older and diffucult time find a new one and hard to get SIP on 7960.
 
 So is there a PTT phone out there that works great with asterisk ?
 
 Thanks so much.
 
 Jerry


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Re: [asterisk-users] How to dial extensions asynchronous-sequentially ?

[Pete Mundy]

Heya Rodrigo

Not sure, but this expansion on Sammy's concept may help you achieve the 
delayed ring on the secondary extensions you were looking for.

exten = _600.,1,Dial(PJSIP/${EXTEN})
exten = _600.,n,Hangup

exten = _600.wait5,1,Wait(5)
exten = _600.wait5,n,Dial(PJSIP/${EXTEN:0:4})
exten = _600.wait5,n,Hangup

exten = 555,1,Dial(LOCAL/6001LOCAL/6002.wait5)
exten = 555,n,Hangup

So you dial '555' and it rings 6001, then 5 second later (assuming 6001 isn't 
answered yet) 6002 starts ringing too (first to answer gets it).

Pete


On 14/07/2015, at 7:24 AM, SamyGo govoi...@gmail.com wrote:

 Anyway here's one way of how I think you can do.
 
 Have a context created to dial the individual user
 
 [dial_user]
 exten = _600X.,1,Dial(PJSIP/${EXTEN})
 ...
 
 and in your code change it to.
 
 same = n,Dial(local/6001@dial_user/nlocal/6002@dial_user/n)
 same = n,Hangup()


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Re: [asterisk-users] How to dial extensions asynchronous-sequentially ?

[Pete Mundy]

Heya Rodrigo

My apologies for the misunderstanding re the delay.

I think the 183 messages problem stems from Asterisk being a B2BUA not a proxy 
and therefore not the tool or this job. But others have more skill around that 
area than I do so please confirm that before accepting it as fact!

Hope you get it resolved. Sorry to muddy the waters :)

Pete


On 16/07/2015, at 9:24 AM, Rodrigo Pimenta Carvalho pime...@inatel.br wrote:


 Hi Sammy and Pete.
 
 Sammy, you are correct. But your example doesn't allow Asterisk forward every 
 SIP 183 message to the caller.
 
 Pete, in fact, I'm not looking for a delayed ring. All extensions must ring 
 at same time. I got  a kind of solution by using:
 
 exten = _6XXX,1,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})}, 60)
 
 However, the Asterisk is rewriting the SDP content of SIP 183 messages, 
 before forwarding it to the caller. That is the new question I would like to 
 solve, because in my project the caller must receive the SIP 183 from callee 
 as it was originally wrote.
 
 Thanks and regards.
 
 RODRIGO PIMENTA CARVALHO
 Inatel Competence Center
 Software
 Ph: +55 35 3471 9200 RAMAL 979   (Brasil)
 De: asterisk-users-boun...@lists.digium.com 
 asterisk-users-boun...@lists.digium.com em nome de Pete Mundy 
 p...@fiberphone.co.nz
 Enviado: quarta-feira, 15 de julho de 2015 18:16
 Para: Asterisk Users Mailing List - Non-Commercial Discussion
 Assunto: Re: [asterisk-users] How to dial extensions 
 asynchronous-sequentially ?
  
 Heya Rodrigo
 
 Not sure, but this expansion on Sammy's concept may help you achieve the 
 delayed ring on the secondary extensions you were looking for.
 
 exten = _600.,1,Dial(PJSIP/${EXTEN})
 exten = _600.,n,Hangup
 
 exten = _600.wait5,1,Wait(5)
 exten = _600.wait5,n,Dial(PJSIP/${EXTEN:0:4})
 exten = _600.wait5,n,Hangup
 
 exten = 555,1,Dial(LOCAL/6001LOCAL/6002.wait5)
 exten = 555,n,Hangup
 
 So you dial '555' and it rings 6001, then 5 second later (assuming 6001 isn't 
 answered yet) 6002 starts ringing too (first to answer gets it).
 
 Pete
 
 
 On 14/07/2015, at 7:24 AM, SamyGo govoi...@gmail.com wrote:
 
 Anyway here's one way of how I think you can do.
 
 Have a context created to dial the individual user
 
 [dial_user]
 exten = _600X.,1,Dial(PJSIP/${EXTEN})
 ...
 
 and in your code change it to.
 
 same = n,Dial(local/6001@dial_user/nlocal/6002@dial_user/n)
 same = n,Hangup()
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Re: [asterisk-users] Asterisk and OSX

[Pete Mundy]

Hi Gents

I thought I'd pop up and add my 2cents when I just happened across this e-mail 
glancing through the list.

I actually do run a production service on Asterisk running on Mac OS X server. 
But of course I do it within a VM environment on Linux (Ie Asterisk on Ubuntu 
under VirtualBox on Mac OS X Server). This gives me all of the niceties of a 
Mac server, along with the ease of administration of Linux for the native 
Asterisk config.

The best part though is that we can easily run up 4 independent Asterisk VMs on 
the one Mac Mini server, then mount it in one of these cases along with another 
one configured the same for hot-standby and we have a fantastic low-power 
(200W) small 1U, 300m-deep rack-mount solution. Stopping the VMs on one Mini 
and launching them on the other (without any change of IPs) can be done in 
9seconds when booted from SSDs.

http://www.sonnettech.com/product/rackmacmini.html

Macs do have their place running Asterisk. Just not natively! :)

Pete Mundy
Technical Director
Fiberphone Limited
Nelson, New Zealand
www.fiberphone.co.nz



On 15/04/2014, at 10:40 PM, Thomas Rechberger t.rechber...@gmail.com wrote:

 Am 14.04.2014 16:19, schrieb Eric Wieling:
 So few people use Asteisk on OSX that I doubt anyone will answer.
 
 Look how many answers he got, i got none
 
 
 
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Re: [asterisk-users] Somewhat OT: Specific SIP packets can cause ethernet controller reset

[Pete Mundy]

On 7/02/2013, at 5:08 AM, Kristian Kielhofner k...@kriskinc.com wrote:

 While not strictly Asterisk related this issue could certainly affect
 some of you:
 
 http://blog.krisk.org/2013/02/packets-of-death.html

A fascinating write-up Kristian. The information it's making it's way around 
the e-mail underbelly of the internet very quickly, and a lot of interested 
(and some very switched on) eyes are seeing it in a short amount of time, or so 
it seems. I first heard about it through a geek friend of mine here in NZ 
before I saw your posting on this list for example.

Good on you for making your research results ( method) public. Well done.

Pete Mundy



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Re: [asterisk-users] Asterisk Messaging Refuses To Work!

[Pete Mundy]

On 31/01/2013, at 8:11 AM, XBrian bobo...@yahoo.co.uk wrote:

 I have changed the dial command to 
 [snip]
 still no joy!

What Dial command are you referring to? There are no instanced of the 'Dial' or 
'Answer' app in any of your examples. Did you read Richard's comment about 
SendText only being valid during an active call? I think he hit the nail on the 
head. If you're not running 'Dial' or 'Answer' then this isn't going to work.

Pete



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Re: [asterisk-users] IVR Menu Sounds

[Pete Mundy]

 Is there a sound package that I need to install?

Have a look in /var/lib/asterisk/sounds - that's where my sounds ended up being 
installed on Ubuntu. If there are sound files in there then you have them 
installed and the next thing to focus on is output from the Asterisk console 
during a fail (it will indicate where to look next).

Pete

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Re: [asterisk-users] Mail list settings?

[Pete Mundy]

On 18/01/2013, at 12:37 PM, Andrew Latham lath...@gmail.com wrote:

 On Thu, Jan 17, 2013 at 6:32 PM, Bryant Zimmerman brya...@zktech.com wrote:
 For some reason the mailing list is sending all messages from the sending
 party.
 This makes it less than ideal when responding; as selecting reply goes to
 the person and not the list.
 Can we have it set back to the old way please?
 
 I just checked back over the list emails and Bryant's email appears to
 be unique in this problem.  I assume it is a simple issue somewhere.
 List admins?

My 2c...

Looking back over recent e-mails, it looks to me like Bryant just has a 
reply-to header on his outbound e-mail's with his e-mail in there. The mailing 
list is simply allowing his address to remain in the 'reply-to' header (while 
adding the list's address too).

I've noticed some others do this too (Chui Kingh Man) is an example, but there 
are others.

So is this a case of the mailing list no longer stripping 'reply-to' headers 
before adding it's own, or is this simply a case of a few users setting 
reply-to when most don't, and those users getting replies directly as well as 
to the list (as one would expect)?

Ie, unless I'm mistaken, it all looks to be operating normally.

But I'd be happy to be proven wrong ;)

Pete Mundy

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Re: [asterisk-users] Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable

[Pete Mundy]

On 17/01/2013, at 4:35 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote:

 Unplug 10.3.22.6, and try pinging it.  If something answers, then you 
 indeed 
 have a clash.  Check your DHCP server configuration, and make sure any 
 manually-assigned addresses are outside its pool of addresses.

If you do this test, remember to make sure to keep pinging with the host 
disconnected for minimum 30 seconds so as to give your local OS's arp table a 
chance to time out (or manually delete the original ARP entry before starting 
the ping).

Pete




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Re: [asterisk-users] Top Posting

[Pete Mundy]

On 30/12/2012, Steve Edwards wrote:

 On Sat, 29 Dec 2012, Don Kelly wrote:
 
 2.   How do we change rule #5?
 
 -1.

+ -1 from me too!

Ie I dislike top-posting on mailing lists and if a democratic approach was 
taken to rule changes (I have no idea is this is the case?) then I would vote 
against the change.

Just my 2c since we're discussing it.

Pete



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Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

[Pete Mundy]

Hi Roy  others.

Sorry for the delayed reply to this thread. The holiday period delayed my 
testing of the A510 and C610 range further, but I have now had a chance to give 
them a little 'ear time'.

On 12/12/2012, at 5:30 PM, Co-op Vacation Rentals coo...@gmail.com wrote:

 Thanks for testing these out.  You said you like the 510 vs 580.  Which one 
 is newer?  I have the 580.  I'm staying tuned for your review of the 610.  
 The 580 isn't DECT compatible and only supports some of the Gigaset handsets.
 I might sell mine and upgrade depending on what you learn about it.
 
 Can you tell me a little more about what you are testing for?
 Quality?
 Simultaneous In/Outgoing Calls using 1 or Multiple SIP accounts.
 Conference Calling?
 Any other Features?

My tests were just for basic functionality of the devices as normal run of the 
mill phones. I wanted to know for example, weather they had any annoying 
'glitches' like the 580's insistence on telling me You have new messages when 
there is only a missed call, or other practical usage issues.

After testing the A510 and C610 my opinion is that I would go for the C610 any 
day. I suspect the A510 is a 'newer' release than the A580 (which seems to be 
EOL now), however I'm not entirely certain about that. The web GUI for the 510 
and the 610 are visually very similar, but both are different to the 580. I 
suspect the 510/610 run a newer shared firmware which has evolved from the 580 
(the 510/610 GUI gives hints of this and it looks overall far more polished 
than the 580).

The 510 handset is quite basic (lower-res mono with coloured backlight). The 
610 handset is visually similar to the 510 (although not the same), but more 
glossy and has a much nicer LCD (colour, higher res). I suspect the 610's 
handset CPU is a little faster too.

I'm surprised to hear you say the 580 isn't DECT compatible. It's manual 
indicates it supports both DECT and GAP, so should be able to take handset 
registrations from other brand handsets (and also register the Gigaset handsets 
with other DECT bases, if desired). Is this not your experience?

I note that the range of other Gigaset handsets compatible with the 580 base is 
a reasonable list too, as shown at hte URL you provided:

 http://gigaset.com/us/en/cms/PageCustomerServicesCompatibility.html


Your link indicates that the C610H handset can be used with your A580OIP base. 
I reckon that would be worth giving a try because the 610 is a nice handset and 
if it works with your bases then it might be a good solution for additional 
phones.

I haven't tested this at all with the 580, but I can confirm that the A510 and 
C610 do happily interoperate with each other (ie, I was able to register my 
A510H with the C610 base and then my C610H with the A510 base). Once the second 
handset was registered with the base, I created a second SIP account login 
definition and assigned the first account to the first handset and the second 
account to the second handset. You can define which handsets receive incoming 
calls on a per SIP account basis and also define which SIP account each handset 
will use for outgoing calls on an individual handset basis.

The configuration options available on each model of base (510/610) appear to 
be identical. This is what makes me think they share a common firmware.

Both handsets do provide notification for missed calls, but thankfully they can 
be configured via the base-station configuration to not flash their MWI led for 
this condition.

What I haven't tried (due to not having any) is registering a standard DECT 
cordless (one that comes with a PoTS analog base) to either of the Gigaset 
base-stations. If that does work OK (fingers crossed), these base-stations 
could be a great upgrade for households with existing analog DECT cordlesses.

Anyway, to wrap up, a note on the other items you asked - I did test 2 
simultaneous SIP calls on separate accounts (and also calling each other via 
the Asterisk server). I only tested with ulaw since that's all we use on this 
server. So note that I haven't tested running 2 concurrent calls using a CPU 
intensive codec; that would be interesting, but out of scope for what I'm 
looking right now. Call quality with 2 ulaw calls in progress is great, just as 
good as a single. I didn't give 3-way conference any decent time testing I'm 
afraid. I did have a shot at it following the instructions in the manual on the 
610 but seemed to only be able to effect a call transfer, not a 3-way bridge. 
But I really didn't give that any decent time and I'm fairly sure that it was 
me doing something wrong there before moving on and not going back.

I have both phones here for another fortnight or so. If there is something 
specific you want tested then let me know and I'll see if I can oblige. 
Otherwise my recommendation is focus on the 610 for now - it's a nice wee phone 
which isn't all that much more expensive, and possibly able to be retro-fitted 
to your existing 

Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43

[Pete Mundy]

On 14/12/2012, at 9:51 AM, Jerry Geis ge...@pagestation.com wrote:

 I did notice one more thing:
 chan_sip.c:17045 handle_request_register: Registration from 
 '5001sip:5001%4010.239.46.200@10.239.46.200' failed for '137.52.88.195' - 
 No matching peer found
 
 Why is there no matching peer I have it defined. I shows in my sip show 
 peers?

I wonder if in fact you have entered into the phone's web GUI the username 
5001@10.239.46.200 when you should have just entered 5001 (with the server 
name being defined elsewhere in the config, eg as the 'domain' value or the 
'proxy' value).

It looks to me as if the phone has encoded the string '5001@10.239.46.200' into 
the username '5001%4010.239.46.200' and then tried to connect to the server 
10.239.46.200 as that user (when in fact you actually want it to simply connect 
as '5001').

Worth trying? Could be a quick fix...

Pete



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Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

[Pete Mundy]

One thing I dislike about the A580H is that the handset always says 'You have 
new messages' if I've missed a call. It wouldn't bug me if it said 'missed 
call' but it tells me I have new messages and even lights up a red LED under a 
button with a picture of an envelope on it.

I'm about to test an A510IP and an A610IP to compare against the A580. Fingers 
crossed neither of them has that issue, because the Gigaset phone is a pretty 
good phone other than that, and the difficulty doing a (blind) transfer, as 
referred to by the OP.

Pete


On 12/12/2012, at 8:57 AM, Roy Abshire r...@coopvr.com wrote:

 I've been using the Gigaset A580 Base and A58H Phone for about 3 years now.  
 Never gave me problems. The call Quality is excellent!
 I only have 1 handset connected to the Base but I want more. I bought a 
 Linksys WIP330 as a 2nd phone to try out and that works just as good without 
 a base unit.
 
 The A580 Base supports up to 6 handsets.
 
 I have 6 Incoming VOIP Numbers using separate SIP accounts pointed to 1 
 Handset but you can point each SIP to separate handsets.
 
 The call goes to the first phone that picks up.  When on a call, picking up 
 another phone makes a separate call and does not conference.  I don't use 
 conference yet but I know you have to put the call on hold or something.
 
 The thing I don't like about the A580 and might be the same on all of them is 
 that you can only specify 1 Sip Account for making outgoing calls.  In other 
 words, all 6 phones would use the same caller id out, but I wanted to be able 
 to choose that because I have a business number and number for each person in 
 our household.  In order to use a different Caller ID (SIP Account) for 
 making outgoing calls I added a extension to my Dial Plan and before making 
 outgoing calls I press *1-6 before the number.
 
 I'm going to try adding more handsets that are compatible.  I want the SL78H 
 but they are so expensive for just home everyday use.
 
 Make sure you check the compatibility page here before buying handsets.
 
 http://gigaset.com/us/en/cms/PageCustomerServicesCompatibility.html
 
 
 Co-op Vacation Rentals
 www.coopvr.com
 15218 Summit Ave
 Suite #300-354
 Fontana, CA 92336
 Phone/Fax (855) 760-COOP (2667)
 
 On 12/11/2012 11:32 AM, sean darcy wrote:
 Siemens A510IP



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Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

[Pete Mundy]

At the risk of continuing off-topic conversation...

Oh the M9 has it's own issues, don't you worry (not to mention it's _way_ more 
expensive than the Gigaset range)!

I've been testing the A510 today and I've decided I like it more than the A580. 
The software (via the web interface) looks more polished and overall thus far 
it has performed fairly well (with no annoying 'new message' statement!). I 
haven't tried multiple SIP registrations with dual handsets yet, but I intend 
to. You guys realise these are standardised DECT handsets right? Ie that's 
where the 6-handset limitation comes from, and you should theoretically be able 
to register other manufacturers' handsets too.

I'm about to try the C610IP model that Kai-Uwe mentioned (over the next few 
days). I think I'll like even more than the A580 since it has a higher 
resolution colour display. And even then it's still miles cheaper than the M9!

Don't get me wrong, I do really like Snom phones too, although the M9 hasn't 
been entirely smooth sailing. I love the 320 desk phone (at the cheaper end of 
the scale, it has a great feel to it).

Isn't it great that they all interact with Asterisk so well (there you go, I'm 
on-topic again ;-).

Pete


On 12/12/2012, at 4:12 PM, Mitul Limbani mi...@enterux.in wrote:

 Mebbe you guys should try snom m9 dect ip phone, i have been using it since 
 over 3 years now without any of these issues.
 
 Mitul
 
 On Dec 12, 2012 4:25 AM, Kai-Uwe Jensen kujen...@gmail.com wrote:
 Using a Gigaset C610IP here, and am very happy with the features. The base 
 station can handle two concurrent SIP calls, and another internal one at 
 that. It does it with a single SIP registration to each server. You can setup 
 multiple servers if you want to and define dial patterns/plans that determine 
 which server gets used. After some playing around with it, I'm now using my 
 setup connected to a single asterisk only. (Let asterisk make call routing 
 decisions based on cost, using an AGI)
 
 Call transfer is working fine, the handsets have a Flash/R key to accomplish 
 this. Using the Flash lets you start a second call, and once answered you can 
 easily conference the second party in (softkey right on the screen), or 
 transfer the call to the other party (via menu, then transfer). Using this 
 capability, someone on a call can easily confer with another party, and 
 bridge them into the call. AFAIK it is not possible for someone to join an 
 existing call easily. You'd have to implement that in asterisk's dialplan, 
 not on the Gigaset phone.
 
 My understanding is that the C610IP has a few more features than the 510. I 
 might've also read somewhere that the 510 is obsolete. Can't find that link 
 right now, but search mgraves.org (use the Gigaset tag to get some initial 
 results).
 
 
 On Tue, Dec 11, 2012 at 2:37 PM, Roy Abshire r...@coopvr.com wrote:
 That is true about the A580. 
 
 I don't like the interface much to check messages.
 
 Besides that every time I go to dial a number...it always uses the first 
 digit pressed to go into phone mode..so I have to press the first digit 
 twice...
 
 I would test other phones but it's for home and I can't fork over $$ to try 
 them all out
 
 I have tested some Nokia cell phones, the N97, N900, and E71 and the E71 and 
 N900 worked well.  I didn't like the N97.
 
 
 Co-op Vacation Rentals
 
 www.coopvr.com
 
 15218 Summit Ave 
 Suite #300-354
 Fontana, CA 92336
 Phone/Fax (855) 760-COOP (2667)
 
 On 12/11/2012 12:52 PM, Pete Mundy wrote:
 One thing I dislike about the A580H is that the handset always says 'You 
 have new messages' if I've missed a call. It wouldn't bug me if it said 
 'missed call' but it tells me I have new messages and even lights up a red 
 LED under a button with a picture of an envelope on it.
 
 I'm about to test an A510IP and an A610IP to compare against the A580. 
 Fingers crossed neither of them has that issue, because the Gigaset phone is 
 a pretty good phone other than that, and the difficulty doing a (blind) 
 transfer, as referred to by the OP.
 
 Pete
 
 
 On 12/12/2012, at 8:57 AM, Roy Abshire 
 r...@coopvr.com
  wrote:
 
 
 I've been using the Gigaset A580 Base and A58H Phone for about 3 years now. 
  Never gave me problems. The call Quality is excellent!
 I only have 1 handset connected to the Base but I want more. I bought a 
 Linksys WIP330 as a 2nd phone to try out and that works just as good 
 without a base unit.
 
 The A580 Base supports up to 6 handsets.
 
 I have 6 Incoming VOIP Numbers using separate SIP accounts pointed to 1 
 Handset but you can point each SIP to separate handsets.
 
 The call goes to the first phone that picks up.  When on a call, picking up 
 another phone makes a separate call and does not conference.  I don't use 
 conference yet but I know you have to put the call on hold or something.
 
 The thing I don't like about the A580 and might be the same on all of them 
 is that you can only specify 1 Sip

Re: [asterisk-users] How to check channel status and move on silently?

[Pete Mundy]

Dear list (FTPer, think I finally spotted one I can help with!)

 I have 10 different routes with few different providers. When I place an 
 international call, I would like the system to try all those routes and place 
 the call through whichever possible. If there is any message but an ANSWER 
 the system should move on to next route. I know this is not the best strategy 
 but there are so many bad routes now-a-days that it's becoming a headache.

I'd recommend looking at the ${DIALSTATUS} Asterisk variable and wrapping up 
some 'Dial' and 'Goto' applications in a macro that calls the first provider 
then looks at the returned ${DIALSTATUS} to make the logic decision of where to 
go from there. Then use the new macro everywhere where you would have 
previously used a single upstream dial.

Further reading references (the second one actually containing an example that 
might be a useful starting point for you to work from):

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

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

Hope this helps!

Pete Mundy






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