[asterisk-users] AUTO: Kevin Larsen is out of the office (returning Mon 01/08/2018)

[kevin . larsen]

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Re: [asterisk-users] Rewrite Outgoing Number

[kevin . larsen]

asterisk-users-boun...@lists.digium.com wrote on 12/14/2017 09:52:32 AM:

> From: "basti" 
> To: asterisk-users@lists.digium.com
> Date: 12/14/2017 09:52 AM
> Subject: Re: [asterisk-users] Rewrite Outgoing Number
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> ok thanks for the answer, i will try it.
> sorry for the question: in which file should it be configured?
> 

In FreePBX, you will set up provider 1 as an outbound route 
(Connectivity/Outbound Routes). You will tell it what dial patterns to use 
that will take that route. You will also specify which trunks to use and 
in what order they should go. One would assume in your situation that you 
want to have provider 1 as your primary trunk and provider 2 as your 
backup trunk should trunk 1 be down.

So, basically, you first need to set up provider 1 and provider 2 under 
Connectivity/Trunks. Make sure that under CID Options you have Allow Any 
CID, otherwise your test won't work. Then you need to set up outbound 
routes under Connectivity/Outbound Routes. Make sure there that Your trunk 
sequence has the Provider 1 trunk as primary. If you want Provider 2 as a 
backup, put that as secondary.

Finally, make sure that under Applications/Extensions, on the General tab 
that you have the Outbound CID set to the number you want to use. That 
will get used whenever you dial out a trunk.

Hope that sets you down the correct path.

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Re: [asterisk-users] Rewrite Outgoing Number

[kevin . larsen]

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208

asterisk-users-boun...@lists.digium.com wrote on 12/14/2017 09:36:06 AM:

> From: "basti" <mailingl...@unix-solution.de>
> To: asterisk-users@lists.digium.com
> Date: 12/14/2017 09:36 AM
> Subject: Re: [asterisk-users] Rewrite Outgoing Number
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> On 14.12.2017 16:30, basti wrote:
> Hello,
> I am new on asterisk and do some tests on freepbx.
> 
> I have 2 SIP provider:
> 
> Provider1: In-/Out- Flatrate, only 1 Number
> Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 
numbers
> 
> On Asterisk site i have 3 phones
> (branch ??, don't know how its called in asterisk)
> 
> Is it possible to do something like:
> 
> Phone 1: Incoming Call: Number1/Provider1 Outgoing Call: 
Number1/Provider1
> Phone 2: Incoming Call: Number1/Provider2 Outgoing Call: 
Number1/Provider1
> Phone 3: Incoming Call: Number2/Provider2 Outgoing Call: 
Number1/Provider1
> 
> I have forgotten an essential thing:
> 
> Phone2 und Phone 3 should use  Line Number1/Provider1 for Outgoing Call
> but show Number1/Provider2 or Number2/Provider2 on caller side.

If, and this is a big if, your provider 1 allows you to use a caller ID 
number that they do not control, then yes, you can do what you want. 

Some providers allow this and some do not. It may be that provider one 
will overwrite whatever you set as caller ID with the number you have 
purchased from them. It may also be that they will allow you to set a 
different outbound caller id. Also, the person receiving the call will not 
know if you have provider 1 or provider 2. It is purely the number and 
possibly a name that they will see.

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[asterisk-users] AUTO: Kevin Larsen is out of the office (returning 07/31/2017)

[kevin . larsen]

I am out of the office until 07/31/2017.

I am out of the office and will have limited contact. For all
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[asterisk13] Multiple transport objects of same protocol in pjsip.conf"
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Re: [asterisk-users] log incoming calls without answering

[kevin . larsen]

> I've already proposed your solution (is the most reasonable) but they
> have more than 60 analogs lines (no faxes) and some of them terminate in
> appliances like alarms, etc, so the solution must not touch in any way
> the connection between the line and his termination: doing a analog to
> digital conversion, passing it to asterisk and the convert it back to
> analog is prone to problems (what if asterisk crashes? or if a gateway
> fail?).
> I can split the existing lines (there are no complex things like adsl or
> digital signaling), convert the branches to digital and terminate then
> into an asterisk machine, so any failure will not affect the old
> circuit, but of course I've to configure asterisk to ONLY LOG calls and
> nothing more.
> 
> This is what they want:
> - line 1 ring
> - line 1 is splitted in two, the first branch (let's say the "analog"
> branch) go to an analog phone, that rings
> - the second branch go through a gateway and then to asterisk
> - asterisk log (with an AGI for example) "line 1 rings at  from 
"
> no more is required from asterisk, if someone answer the analog phone or
> not is not my business.
> 
Ok, so I would agree with them that a conversion to digital and back again 
would tend to break things like fax lines and alarm lines. My analog lines 
in my facilities are there because a lot of alarm systems just don't work 
with SIP at all. It's something the alarm companies are going to have to 
figure out in the next decade or so as the Telcos are moving away from 
copper and switched networks and towards fiber and packet based networks.

I honestly don't know if you can do what you want without some piece of 
equipment picking up the line. What I would do is get an analog line, an 
analog phone, an analog to sip device (there are many to choose from) and 
a basic asterisk instance. I would then make a small test setup where the 
analog line goes to a splitter. One side of the splitter goes to your 
analog phone. One side goes to your analog to SIP converter and then into 
your asterisk instance via your ethernet network. Use your cell phone to 
call the number of your analog line and see if it works. You would have to 
code a basic dialplan on the asterisk side and set up the trunk to your 
converter, which I am assuming you know how to do.

This would at least give you a fairly low cost way to test to see if you 
can trigger what you want on the Asterisk side without also triggering the 
line itself to be answered. I would also note that you would only be able 
to log incoming calls this way. I can't see a way you would be able to 
detect an outgoing call from the analog extension.

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

[kevin . larsen]

> From: Fabio Moretti 
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Date: 04/20/2017 03:26 PM
> Subject: [asterisk-users] log incoming calls without answering
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> 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

This gets kinda Rube Golberg-ish, but convert the incoming analog line to 
sip, route it through asterisk and have asterisk do its thing before 
converting it back to analog to send to the phone. Only problem is you get 
a lot of extra hardware involved in the mix to make it work. It will be a 
lot of expense and trouble, so you need to make sure that whatever part 
you want asterisk to play is worth that effort. Also, I wouldn't touch a 
fax line in this manner.

If you could give a bit more info on what you want asterisk to do, we 
could maybe give better advice on how to solve your problem.

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Re: [asterisk-users] Cisco IP 8841 asterisk integration

[kevin . larsen]

> True agree, problem is somehow the people purchased am 
> supporting to overcome that. Trying level best... around 20 
> phones has been purchased

Ah, yes, the "we purchased these without consulting you, but it is up to 
you to make them work" school of thought. It often goes with, "Well, what 
are we paying you for?" and "It's a phone, it shouldn't take you long to 
make it work."

I have to say, unless I am working with a Cisco phone system, Cisco phones 
are not my favorite beasts to work with.

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Re: [asterisk-users] FAX CNG detected but no fax extension

[kevin . larsen]

> Hello,
> I have a question regarding incoming fax to local file (on the 
> Asterisk server).
> While the fax is received properly (I have the tiff file generated 
> as expected) I get the warning 'FAX CNG detected but no fax 
> extension' on the consol.
> 
> If the fax is received ok then what 'fax extension' does it expect 
> and what should I do there? 
> 
> My Setup:
> Sender -> Public PSTN -> provider -> SIP trunk (configured with 
> G711a) -> Asterisk (13.6.0)
> 
> My extension.conf on relevant section is this (obviously this is not
> production code):
> exten => s,1,Answer()
> same =>  n,Verbose(0, Attempt to Receive FAX)
> same =>  n,Set(FAXOPT(gateway)=no)
> same =>  n,ReceiveFax(/var/workspace/testfax.tiff,d)
> same =>  n,Hangup()
> 
> and 
> Server*CLI> module show like fax
> Module Description 
> Use Count  Status  Support Level
> res_fax.so Generic FAX Applications 
> 1  Running  core
> res_fax_spandsp.so Spandsp G.711 and T.38 FAX 
> Technologies  0  Running  extended
> 2 modules loaded

The good news is you don't really have anything wrong and as things are 
working as expected, you can ignore the warning if you so choose.

What generates that error is that on your trunk, you have faxdetect=yes. 
This will cause Asterisk to listen in to all your calls on that trunk and 
try to detect a fax and if it finds it will redirect it to a fax extension 
to be handled as a fax.

You have written a fax handler for your fax lines, but that doesn't stop 
the fax detection from trying to route it to an extension called fax. 
Since this doesn't exist in your case, you get the warning, but the fax is 
received because you are handling in the current path.

Where things would actually break is if someone sent a fax to one of your 
voice lines. If you don't have a fax extension to send it to, the person 
being called would pick up to fax tones. If you do have a fax extension, 
they would get the call yanked from them and it would be sent over to the 
fax extension. In my particular case, testing shows I get about half a 
ring to my desk phone before the system determines fax call and sends it 
to the fax system.

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Re: [asterisk-users] Problem "re-parking" calls

[kevin . larsen]

> All;
> I have a problem with regards to “re-parking” calls and I was 
> hoping someone could shed some light on the topic. Consider this 
scenario:
> 
> (1) An inbound call comes in and the attendant answers it
> (2) The attendant places the call on hold and the caller is sent to 
> extension 701
> (3) Blah, blah, blah. The attendant does something and tells John 
> Doe to pick up the call on extension 701
> (4) The attendant then picks up the call on 701 and tells the person
> that John Doe will be right there to help them
> (5)  The attendant then re-parks the call but now the caller is sent to 
702
> (6) John Doe can't find the call anymore
> 
> 
> Is there something obvious that I am missing? Has anyone else found 
> this to be a problem? Any insight at all would be greatly appreciated.
> Regards;
> John V.

Your problem occurs in step 4 & 5. I don't believe that you can pick up 
the call and then ever be guaranteed to get the same parking position when 
you put it back in park. What would happen if someone else parked a call 
in between steps 4 and 5 and they got 701 because it was free. Once 
parked, the call should remain so until it is picked up or times out back 
to the attendant.

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Re: [asterisk-users] Asterisk inside network. What phone works well?

[kevin . larsen]

> I have Asterisk running well inside our network. I did some 
> experiments exposing it to internet but had some issues:
> 1. NAT issues (voice one way, etc). From what I understand double-
> NAT users will always have something like this
> 2. Immediately I see people trying to hack into. I did configure 
> Fail2Ban and it works somewhat, but not 100%. Erroneous logs, etc
> 
> So.. I ended up closing network. Currently most users inside 
> network. My home router have GRE tunnel to office so phone works just 
fine.
> Another user uses VPN and soft phone and it works good too.
> 
> Now I need to setup some users with actual phone devices and none of
> those solutions will work. So, I did some research and found 
> that some phones have VPN capability built in. 
> 
> Right now I use Cisco SPA504G phones. We have auto-provisioning for 
> them, works well. But I don’t think they have VPN capability.
> 
> 
> What I found it that Cisco 525g2 has AnyConnect functionality (SSL 
> VPN) but not sure if this is what I need.
> 
> We have Mikrotik router. Can I setup VPN on router and have this 
> Cisco phone auto-dial VPN and then connect to Asterisk? I’m asking 
> to see if this will work before I go in and buy that phone.
> Or maybe there is other devices/solutions you suggest? I’d like to 
> stay with Cisco because I’m somewhat familiar with provisioning those..

I haven't done this myself, but I think what you need to look at is phones 
that can do IPSEC vpn setups.

For the Mikrotik router, this may be helpful to start investigating:
http://wiki.mikrotik.com/wiki/L2TP_%2B_IPSEC_between_Mikrotik_router_and_a_PC

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[asterisk-users] AUTO: Kevin Larsen is out of the office (returning 09/06/2016)

[Kevin Larsen]

I am out of the office until 09/06/2016.

I am out of the office and will have limited contact. For all
emergencies/issues, please contact the helpdesk at
helpd...@pioneerballoon.com or 316-688-8777.


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Need ISDN call generator" sent on 8/29/2016 2:58:18 AM.

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Re: [asterisk-users] Getting better Caller ID

[Kevin Larsen]

> Hello,
> 
> We use Asterisk and as per book we use MAC addresses as user names.
> So, when call coming in from outside (SIP trunk) - caller id is good.
> 
> But when users calling each other on extensions - they see MAC 
> addresses. How would I make it so we see actual names instead of MAC
> addresses? Without changing users..
> 
Do you have a line like the following in your sip.conf for each user?

callerid="Name Here" 

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Re: [asterisk-users] open source pbx free

[Kevin Larsen]

> Anyone have any experience running an open source pbx and call 
> center solution?Need to start a call center of 10 users  and i need help 
 
> 
> I have already  installer a server with Ubuntu Server 14.04  , E1 
installed 
> 
> Please advice me how to process  from here 
> 
> Regards 
> 
> Yves 

Many of us on this list have experience running call centers off of 
Asterisk, myself included. If you haven't done Asterisk before, you might 
want to bring in some outside help in order to smooth over the process. It 
isn't that you can't do it on your own, but expect there to be something 
of a steep learning curve. If you haven't had experience with VOIP before, 
you will run into issues that you didn't even know were possible, and in a 
call center scenario, you will have people breathing down your neck 
wanting things fixed/changed.

The great thing about Asterisk is that if you know what you are doing, you 
can pretty much bend it to your will. It isn't perfect (no software is), 
but there have been very few requests from end users that I haven't been 
able to fulfill once I understood what they really wanted. Phone systems 
are big and scary and hard for technical people. Most non-techies don't 
know enough about them to even know the right questions to ask. That's why 
your very first job is to find out what does the client really want/need 
their phone system to do. Call center of 10 users gives you a direction to 
go in, but it isn't enough to design the phone system. You need to find 
out what exactly do they want to happen when a call comes in. How should 
it be routed. Are they going to use call queues? By indicating a call 
center it is likely they will, but I have seen it where they don't.

Once you have your requirements mostly decided, then you can go ahead and 
decide on what to do next. If it will fit the bill, especially for a new 
asterisk user, there are many prebuilt distributions that will make 
setting up and maintaining your Asterisk solution easier. They have nice 
web interfaces to handle all the heavy lifting. As you sound pretty new to 
VOIP, this may be the way you want to go. If they don't meet your needs, 
then you may be into custom programming the dialplan and gets a lot more 
involved. 

Good luck and enjoy the journey. Also, the more specific you can make your 
questions, the better and more likely the fine folks on this board will be 
to respond with helpful information.

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Re: [asterisk-users] Is possible to use FXO Digium card like a Fax modem?

[Kevin Larsen]

> There are also cheap USB fax modems that you can attach to an FXO 
> port and that works fine. All you have to do then is configure 
> asterisk to detect incoming faxes and route them to that port 
> (faxdetect=yes?).
> 
> This worked great for me when I had all my incoming calls coming 
> over a Century Link POTS line. As I approach retirement and want to 
> save money, I switched from the $44/month POTS line to a pennies-
> per-month VOIP service via IAX registration. So now I'm wondering 
> whether this setup would still work. The question undoubtedly shows 
> my ignorance of telephony stuff. I'm willing to do my homework, I 
> just want to know if it's even possible to do this, or if there are 
> better ways to handle fax over VOIP.

I am going to say this with tongue only partially in cheek. The better way 
to do fax over VOIP is not to do it. It is finicky and unless you have a 
real need for it, it isn't worth the time it takes to make it all work. 
Even working, you still have complaints every time a fax fails to send or 
receive as people somehow have this expectation that faxes should never 
fail. To quote the movie War Games, "The only winning move is not to 
play."

It would be preferable to use a scanner and email to send documents if at 
all possible. If you still need the occasional fax, I would recommend 
using a fax service and letting that be someone else's headache.

That said, my company still has plenty of people who insist that faxes are 
the greatest thing since sliced bread, so I get the fun of supporting 
them. Your options, depending on scale are to use one the solutions you 
can integrate right into the Asterisk server or to use an external package 
and then you just forward the calls from your asterisk box over to your 
fax software (this is the one I use).

Make sure that your SIP/IAX provider supports T.38 faxing (specifically 
transcoding) as this will make your life much easier. You have to be 
careful here as many providers will happily pass T.38 along if it comes in 
that way, but if someone with an analog line/fax setup sends you a fax, it 
will hit their system as audio and pass on to you as audio, which with SIP 
can be fraught with danger unless you have a really excellent connection 
to your sip provider. With transcoding, they can convert it as it enters 
their system to T.38 and then just pass the T.38 to you, which results in 
greater successes. T.38 passthrough is common, transcoding less so, but it 
is getting more common as time goes on.

Also, if your provider does not support T.38 transcoding, plan on sticking 
with ulaw or alaw for faxing. The compressed codecs do not allow the audio 
signal to pass properly and faxes will not work. 

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Re: [asterisk-users] what to do when a sip password includes a semicolon

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 03/11/2016 01:43:47 PM:

> From: Saint Michael 
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> , 
> Date: 03/11/2016 01:44 PM
> Subject: [asterisk-users] what to do when a sip password includes a 
semicolon
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> ​I got a new sip account, and the format
> register=> user:passwrd@proxy:port
> fails when the sip password ​has a semicolon
> Is there a possible workaround?
> I cannot change the password, it comes from the provider.

Try escaping the semicolon with a backslash. A password of abc;123 would 
become abc\;123
Not entirely certain that would work, but it would be the first thing I 
would try.

Also, I think a provider would be amenable to changing a password if it 
was problematic for some reason, but try the backslash first.

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Re: [asterisk-users] 2 devices same *actual* extension - can it be done

[Kevin Larsen]

> Can someone tell me if this is possible?
> 
> I currently have a VOIP phone registered on an Asterisk PBX at a 
> remote location (working fine).
> I want to install an Asterisk PBX at the local location. I will be 
> porting the current POSTS lines to SIP trunking.
> So now I want the remote line and the local lines to appear on the 
> same handset.
> This would mean I would have to pass internet to the phone for the 
> remote extension and also register the local extensions.
> So, for example, I could have the remote extension assigned to line 
> one (ACCOUNT 1 on the Polycom handset), and the local extensions 
> assigned to lines two, three, and four ( ACCOUNTS 2,3,4).
> 
> How do I do this?
> 

So, the first thing you will have to do is to make sure that your phone 
has routes to and can talk to each pbx over the network. Depending on your 
network design, this may be pretty simple or it may get pretty complex and 
will be hard to give a definitive answer in this discussion without more 
details. A good test might be to see if the phone can ping the pbx. Since 
you specifically mentioned a Polycom handset, look under 
Menu-Status-Diagnostics-Network-Ping. This will possibly help you to know 
that you can reach the pbx from the phone (provided your network is set up 
correctly and the pbx responds to pings). Note, many network designs will 
actually block pings even when the SIP and RTP traffic will traverse it 
just fine, so a failure here isn't necessarily the kiss of death.

Next, you will need to set up your phone to register with each PBX. 
Polycom has excellent docs on how to perform a setup using xml 
configuration files. Here is an example with four lines connecting to four 
different voip servers on a Polycom phone. Please note that I do not 
endorse the insecure usernames and passwords used here. They don't follow 
best practices and are only here for an example.



Note that this is just one small section out of a much larger 
configuration file used to completely configure a Polycom phone. Assuming 
you have the rest of your configs working, this would then put 4 lines 
onto the phone, each pointing to a different pbx and each labeled 
uniquely.

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Re: [asterisk-users] Passing Caller ID through Digium Gateway

[Kevin Larsen]

> Hi All,
> 
> I've setup a Digium G100 VoIP gateway to replace an internal PCI VoIP 
> card in our Asterisk PBX.  When using the VoIP card the callerid entries 

> listed in sip.conf were displayed when calling someone over the PSTN. 
> Now, however, though the gateway it just displays the default number 
> assigned to our PRI.  I'm wondering if anyone having experience with the 

> Digium gateways can point me in the right direction to have the gateway 
> respect the callerid entries listed in sip.conf.  We are using an older 
> Asterisk 1.6 build.

We use G100 and 200s at a few of our sites and caller id passes through 
just fine. Check under Configuration -> call routing rules to make sure 
you don't have a caller ID name and number set in there. You have to take 
it off of Simple Entry Mode to see the options.

Also, on your SIP Endopoint configuration, under the call settings tab, 
make sure you have your Caller ID Presentation set correctly. From the 
help on that option:

Caller ID Presentation:Handles Caller ID presentation on outgoing calls. 
Allow for prohibiting Caller ID presentation, and defines whether the 
information has been screened by an authoritative source. Options other 
than screening, allowed, and prohibited indicate that the Caller ID was 
provided by the network. 

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Re: [asterisk-users] Queue logfile txt format in mySQL needed

[Kevin Larsen]

> From: Thomas 
> To: asterisk-users@lists.digium.com, 
> Date: 01/21/2016 04:17 AM
> Subject: [asterisk-users] Queue logfile txt format in mySQL needed
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> Hello,
> 
> Iam using queues and agents, thats OK.
> 
> I have interesting information form Asterisk in txt file format
> var/log/asterisk/queue_log
> 
> Today Iam reading these txt files and wrote them in an mySQL databases.
> 
> I would need this information more realtime. Some information I do 
writing in 
> the dialplan direct in an mySQL database.
> 
> Is there any way that Asterisk write this information direct in an mySQL 

> database instead of using var/log/asterisk/queue_log?

I haven't done this myself, but it looks like you just need to set up the 
appropriate database connections. See here for a semi-recent example:
http://stackoverflow.com/questions/30161384/asterisk-11-queue-log-to-mysql

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Re: [asterisk-users] Forwarding call if extension busy

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 01/04/2016 08:55:40 AM:

> My question:
> 
> - two extensions:  and 
> - an active call on 
> - incoming calls to  should be forwarded to  (call advice!) and 

> 
> I know how can I forward an incoming call to more than an extension, 
> but I have no idea how can I get the information, that  has 
> already an active call...
> 

I am not sure if I completely understand what you are trying to do, but it 
sounds like you want to query the DEVICE_STATE function.

For instance, my customer service department has this thing against ever 
having their phone ring a call while they are already on a call, so for 
these special little snowflakes, I have  the following line:

same => n(voice),GotoIf($["${DEVICE_STATE(sip/${EXTEN})}" != 
"NOT_INUSE"]?voicebusy)

Basically, this little line looks at the extension and if it shows 
anything other than free (NOT_INUSE), it jumps to the voicebusy line in 
the dialplan. The voicebusy line just hits voicemail directly.

You can use this same idea to branch your logic and handle a variety of 
situations. In my case, I only want to actually perform the dial if the 
phone is currently not in use, so my logic was fairly simple.

See here for reference:
https://wiki.asterisk.org/wiki/display/AST/Device+State

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Re: [asterisk-users] Forwarding call if extension busy

[Kevin Larsen]

> Kevin Larsen <kevin.lar...@pioneerballoon.com> schrieb:
> 
> > I am not sure if I completely understand what you are trying to do, 
but it 
> > sounds like you want to query the DEVICE_STATE function.
> 
> IT WORKS
> 
> Thank you very much!
> 

Glad I was able to help. You are most welcome.

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[asterisk-users] AUTO: Kevin Larsen is out of the office (returning 10/24/2015)

[Kevin Larsen]

I am out of the office until 10/24/2015.

I am working in Mexico with limited availability. If the matter is urgent,
please contact the Pioneer Helpdesk.


Note: This is an automated response to your message  "Re: [asterisk-users]
Live Recording on the NAS?" sent on 10/15/2015 1:55:13 PM.

This is the only notification you will receive while this person is away.


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Re: [asterisk-users] Asterisk => Mediant 1000 (AudioCodes) => PSTN (E1)

[Kevin Larsen]

> 
> Does anyone have any information for me?
> 
> 
> Welinghton.
> 
> 
> 
> Citando Welinghton Magno Guimaraes :
> Hello!
> 
> I am setting up an Asterisk server with a Mediant 1000 (Audiocodes) 
> to make external links. Does anyone have any manual or instructions 
> on how to proceed?
> 
> Asterisk  =>  Mediant 1000 (AudioCodes)  =>  PSTN (E1)
> 
> 
> I will be very grateful for the help.
> 
> Thanks!
> 
> 
> Welinghton.
> 

If you do a search for mediant 1000 asterisk you will find some pages that 
might help. One of the problems I have found (I have a couple of 
AudioCodes devices), is that they do not publish anything resembling a 
useful manual to assist end users in setting up their devices. They want 
you to pay for a support contract and for install services instead.

I figured mine out by a lot of trial and error, unfortunately. My devices 
were for fxs/fxo, so unfortunately I doubt me experience would be much 
help.
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Re: [asterisk-users] Share agents state?

[Kevin Larsen]

> Is it possible to share all agents state? if an agent is on the 
> phone on a queue on one of the Asterisk servers, other servers will 
> need to about it and therefore, will be able to operate adequately?
> For instance, an agent is a member of two queues (app_queue 
> realtime) and those queues on separate server.
> Thanks 

You can indeed share a phone's state between servers. If using chan_sip, 
you will be looking at doing something like XMPP. If you are doing pjsip, 
you can do it directly without needing the xmpp server. 

For pjsip, look at 
https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP

For chan_sip, look at 
https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMPP+PubSub

For the record, I have done the xmpp setup and it works well, but there 
was a pretty steep learning curve involved in getting everything working. 
I haven't had a chance to look at upgrading to Asterisk 13 and pjsip to 
set it up, but the configuration looks to be much easier.

I use it because I have Site A which hosts a customer service call queue 
where most of the agents exist on the Site A server. However, I have two 
agents who are at Site B and we don't want to send them a call from the 
Site A queue if they are already on a call from the Site B server. Seems 
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Re: [asterisk-users] How to integrate Asterisk with XMPP

[Kevin Larsen]

> 
> How to integrate Asterisk with XMPP ?
> 

What you are asking for isn't a simple question to answer. What exactly do 
you want to accomplish by integrating XMPP? Shared states among multiple 
extensions? Passing messages between extensions? Depending on what you 
want and what infrastructure you have in place will all influence the 
answer.

Also, you will get better responses if you say what you have tried and 
what isn't working or say what you goal is and ask for pointer on how to 
get there. Depending on what you want to do, there are multiple tutorials 
available online, but I will say that I did find it was a bit of trial and 
error to get xmpp working in my organization. I use it for allowing 
extensions on remote sites to join in to some of our call queues, thus 
needing our (multiple) asterisk boxes to be able to share extension states 
with each other. It wasn't the easiest thing in the world to get working 
on the 11 series. 

Depending on what you want to do, the new pubsub features in PJSIP in 
Asterisk 13 series may do what you want. I know I am looking forward to 
investigating them and quite possibly getting rid of my xmpp setup.

https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP-- 
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Re: [asterisk-users] Receiving faxes with spandsp question

[Kevin Larsen]

 I’m trying to add fax functionality to my asterisk installation. 
 Right now I’m focusing on receiving faxes. This is not explained in 
 a book, but I assume that I can use same context, add “fax” 
 extension and if someone calls to send fax - it will autodetect. Right?
 
  Per book, I made following setup additions:
 
 1. In sip.conf [general] I added:
 
 ;FAX stuff
 faxdetect=yes
 t38pt_udptl=yes
 
 2. In extensions.conf I hade something like this:
 
 [from-callcentric]
 exten = s,1,Goto(automated_attendant,s,1)
 
 ; FAX handling stuff AS IN BOOK
 exten = fax,1,Verbose(3,Incoming Fax)
 same = n,Set(FAXDEST=/tmp) ; folder where faxes will be 
stored
 same = n,Set(tempfax=${STRFTIME(,,%C%y%m%d%H%M)})
 same = n,ReceiveFax(${FAXDEST}/${tempfax}.tif)
 same = n,Verbose(3, - Fax receipt completed with status: $
 {FAXSTATUS})
 
 Well, that didn’t work. Trying to send fax - it was going to my 
 autoattendant and never triggered fax. So, I made a change like so:
 
 3. Changed extensions.conf
 
 [from-callcentric]
 ; FAX handling stuff AS IN BOOK
 exten = s,1,Verbose(3,Incoming Fax)
 same = n,Set(FAXDEST=/tmp) ; folder where faxes will be 
stored
 same = n,Set(tempfax=${STRFTIME(,,%C%y%m%d%H%M)})
 same = n,ReceiveFax(${FAXDEST}/${tempfax}.tif)
 same = n,Verbose(3, - Fax receipt completed with status: $
 {FAXSTATUS})
 
 I just made it fax handling context, and I got FAX :)  But, 
 while fax was received I was getting following:
 
 [2015-06-24 23:40:28] WARNING[47369][C-0005]: res_fax_spandsp.c:
 438 spandsp_log: WARNING T.30 ECM carrier not found
 
 
 QUESTIONS:
 
 1. Should I do something about this warning?
 2. How do I receive fax and have main entry to auto attendant in a 
 same context? Can I have it on same puplic phone number?
 

I think your problem may be that even though you created the exten = fax 
line, it never has a chance to auto detect and go there as it has already 
left that context before it has detected the fax and then has no fax 
extension to redirect to. You could put your fax extension in the 
automated_attendant context and that should work. I recommend a slightly 
different way of handling faxes.

What I did was create an incoming fax context (fax_incoming). In your 
above example, that is where the fax extension would live. That way I can 
handle my reception of faxes in one spot and if I ever need to bug 
fix/change my dialplan, I only have to do it in one spot. Then anywhere 
that I want to autodetect faxes and move them to the fax context I put the 
following extension code:

exten = fax,1,Goto(fax_incoming,${dialednumber},1)

Of course, if you don't want the comment in there, that could be reduced 
to just one line. Also, ${dialednumber} is just a variable I use to hold 
the originally dialed number in case it has been altered as it goes 
through my dialplan so that I can have my CDR records show what was 
originally dialed in case I need to go back later. In your example, you 
would replace ${dialednumber} with whatever you need to work with your fax 
handler. I have multiple fax numbers, so I like to know which one was 
dialed to reach that spot. Makes bookkeeping easier.

I have this working on my sites that use an IVR and based on the timing, 
it gets a few seconds into playing the ivr message usually before it 
detects the fax and redirects it to the proper fax context. I have 
separate fax numbers, but this does catch those who don't pay attention 
and dial the main number instead of the fax number. I also use it on the 
direct dials to my phones. When I get a fax that way, my desk phone will 
get about one ring before the fax is detected and the call is moved away.

Faxing can be finicky to get working how you want it, but you can usually 
make it handle the faxes like you want.
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Re: [asterisk-users] asterisk email to fax

[Kevin Larsen]

 Since the O.P. said he's using it for his home office, I think he'll
 be able to control user expectations :-)


I provide tech support to my parents on all their computers. The amount of 
annoyance I have dealt with in the last few months over the fact that a 
recipe program and various card making programs designed for Windows 
3.1/95 won't run on my mom's Windows 7 64 bit computer tells me you are 
not as right as you think you are.-- 
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Re: [asterisk-users] FXS Solutions for modems and other non jitter tolerant devices

[Kevin Larsen]

 The legal and medical communities still seem to prefer faxing, in 
 the ( mistaken? ) belief that it is more secure. In fact the medical
 community is fearful of the legal beagles.
 
 These groups are really slow to change.
 At least in the USA

The couple of times I have received medical faxes to my fax bank scare me 
about the actual security. My company is not in the medical field, nowhere 
close, in fact.

In one case, the fax included the patients name, address, phone, Date of 
Birth, SSN, and confidential medical history. The comment I made to a 
coworker was that if I wanted to steal an identity, they had just handed 
me everything I would need.

In the second case, it was a question from a pharmacy to a doctors office. 
Not quite so bad. I called up the pharmacy and said I had a problem with a 
fax they had sent. After asking me for some information from the fax so 
they could identify which patient I was calling about they asked what the 
problem was. I replied that I was a manufacturer of balloons and not a 
doctor's office. To say there was a bit of panic creeping into the guys 
voice on the other end was an understatement. I think I triggered some 
HIPAA reporting provisions.-- 
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Re: [asterisk-users] small homebrew pbx

[Kevin Larsen]

 I don't know this 'translates' to Italy, but this is what I would advise 

 somebody in the US to consider, assuming you have a reliable Internet 
 connection.
 
 0) I hope you mean you want to run Asterisk at home instead of 'Asterisk 

 at Home.' A@H was an ancient distribution from around 2005.
 
 1) Rent a DID (a 'PSTN number') from a reputable SIP provider. This 
 eliminates the need for a PCI/USB interface and you won't disrupt your 
 'business' while you figure out how to configure and test your Asterisk 
 server.
 
 In the US, you can rent a DID for about $1.50 per month and about a 
$0.01 
 per minute of 'talk time.' For 10 calls per day, this should beat the 
hell 
 out of a 'landline' monthly standing fee.
 
 In the US, it costs less than $20.00 to 'port' your existing number if 
you 
 are really in love with it.
 
 2) Ditch the 'room warmer' and find something really small and cheap to 
 run. I live in San Diego and we pay $0.32 per kWh. I'd guess running 
your 
 rig would cost me $50.00 to $100.00 per month just in electricity -- and 

 probably that much again in the summer for additional Air Conditioning.
 
 Take a look at Soekris net4801. It's pretty old (but very reliable) and 
 it's CPU will limit you on what OS you can run, but it will give you an 
 idea of how small (and cheap to power) an 'Asterisk server' capable of 
 handling a couple of simultaneous calls can be.
 
 For a more modern server, look for something small and cheap based on 
 something like an Atom processor. Maybe a used laptop. If the battery is 

 still good, you've solved your UPS problem as well. Although, if you 
lose 
 power, you've probably lost your Internet connection as well so you 
could 
 only make calls between extensions.
 
 3) For the IP phones, check out ebay.com. Last year, I picked up 3 
Polycom 
 SP 501's for $20.00 each. A little dated, but a great phone.

I gotta agree with most all of this. Asterisk has been shown to run on a 
Raspberry Pi and the Raspberry Pi 2 and will handle a few simultaneous 
calls. Another resource is http://www.plugpbx.org/

For home use, I would think either would be a good low power way to run 
Asterisk. Unless you just really need the land line, ditch the analog line 
and go voip from start to finish.
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Re: [asterisk-users] Am I cracked?

[Kevin Larsen]

 Very strange...
 I ran the Asterisk CLI for other tasks, and suddenly I got this message:
 
   == Using SIP RTP CoS mark 5
 -- Executing [000972592603325@default:1] Verbose(SIP/192.168.
 20.120-002a, 2,PROXY Call from 0123456 to 000972592603325) innew 
stack
   == PROXY Call from 0123456 to 000972592603325
 -- Executing [000972592603325@default:2] Set(SIP/192.168.20.
 120-002a, CHANNEL(musicclass)=default) in new stack
 -- Executing [000972592603325@default:3] GotoIf(SIP/192.168.20.
 120-002a, 0?dialluca) in new stack
 -- Executing [000972592603325@default:4] GotoIf(SIP/192.168.20.
 120-002a, 0?dialfax) in new stack
 -- Executing [000972592603325@default:5] GotoIf(SIP/192.168.20.
 120-002a, 0?dialanika) in new stack
 -- Executing [000972592603325@default:6] Dial(SIP/192.168.20.
 120-002a, SIP/pbxluca/000972592603325,,R) in new stack
 [Jun  8 21:42:50] WARNING[18981]: app_dial.c:2345 dial_exec_full: 
 Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
   == Everyone is busy/congested at this time (1:0/0/1)
 -- Executing [000972592603325@default:7] Hangup(SIP/192.168.20.
 120-002a, ) in new stack
   == Spawn extension (default, 000972592603325, 7) exited non-zero 
 on 'SIP/192.168.20.120-002a'
 [Jun  8 21:43:22] WARNING[16633]: chan_sip.c:3830 retrans_pkt: 
 Retransmission timeout reached on transmission 
 8dc31ca4e660a0408450715638784d86 for seqno 1 (Critical Response) -- See 
 https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
 Packet timed out after 32001ms with no response
 
 At the time no phone try to call...
 On my Firewall I see a SIP packet coming from an IP in Palestine...
 Am I cracked? I think I disabled all guest access. How can I check if 
my
 Asterisk allows guest to originate calls?

Based on SIP packets coming in from IP addresses you don't recognize, 
while you may not be hacked, you would seem to have people probing your 
system. One thing you can do at the firewall level is restrict inbound sip 
communications to only those from your external phone providers. Depending 
on their setup, they should be able to give you an IP, a range of IPs or a 
name that can be used (i.e. sip.myphoneprovider.com). If you restrict your 
inbound sip to that, it will be very helpful. Also, there are further 
steps you can take to harden your systems. An internet search will bring 
up many, but here are a couple of good ones:

http://blogs.digium.com/2009/03/28/sip-security/
http://www.ipcomms.net/blog/70-11-steps-to-secure-your-asterisk-ip-pbx
http://nerdvittles.com/?p=580-- 
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Re: [asterisk-users] Am I cracked?

[Kevin Larsen]

 OK, I set alwaysauthreject = yes and I discovered a allowguest, which I 
set
 to no, too.
 The PBX is behind a Firewall and I just allow UDP 5060 and 1-10100.
 Now I log the SIP-pakets coming from Internet, too...
 
 Hopefully I solved my problem...

Make sure you have solved the problem. You don't want to get hit with a 
phone bill for calls from your location to Israel. Basically, they are 
hoping that you are running the equivalent of a mail server open relay. 
They are trying to use you to dial out to another number. You don't want 
to pay for these calls.

The calls are being dumped into your default context. It's not matching on 
your gotoif statements, so finally it is trying to execute this:
Dial(SIP/192.168.20.120-002a, SIP/pbxluca/000972592603325,,R) in 
new stack

Not sure what trunk pbxluca is, but if that is an outbound trunk, then 
this is very bad. The only reason it would fail then is if they have the 
outbound dial pattern wrong, which is a sure sign that you are open in the 
future to having someone make this kind of call in a way that does work 
and leaves you on the hook. Based on your email address, I am guessing you 
are in Germany. Looks like they almost have the correct outbound pattern 
for dialing from Germany to Israel. It should be 00972592603325 (notice 
the one less zero in the front). Please tell me that pbxluca is not an 
outbound dialing context? If it is, you need to fix this very quickly.-- 
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Re: [asterisk-users] Am I cracked?

[Kevin Larsen]

  Make sure you have solved the problem. You don't want to get hit with 
a 
  phone bill for calls from your location to Israel. Basically, they are 

  hoping that you are running the equivalent of a mail server open 
relay. 
  They are trying to use you to dial out to another number. You don't 
want 
  to pay for these calls.
 
 Of course, but how can I test, if I am an open relay?
 
  The calls are being dumped into your default context. It's not 
matching on 
  your gotoif statements, so finally it is trying to execute this:
  Dial(SIP/192.168.20.120-002a, SIP/pbxluca/000972592603325,,R) 
in 
  new stack
  
  Not sure what trunk pbxluca is, but if that is an outbound trunk, then 

  this is very bad. The only reason it would fail then is if they have 
the 
 
 This is one of my outbound trunk...
 
  outbound dial pattern wrong, which is a sure sign that you are open in 
the 
  future to having someone make this kind of call in a way that does 
work 
  and leaves you on the hook. Based on your email address, I am guessing 
you 
  are in Germany. Looks like they almost have the correct outbound 
pattern 
  for dialing from Germany to Israel. It should be 00972592603325 
(notice 
  the one less zero in the front). Please tell me that pbxluca is not an 

  outbound dialing context? If it is, you need to fix this very quickly.
 
 How can I fix it? Of course, I need to be able to call any phone on this
 world...
 On a Mail-Server I'd restrict outgoing calls to authenticated users. I 
was
 sure, that Asterisk already do that, but I'm not sure anymore...
 How can I restrict it?

I am sure others can chime in, but first things first, you want inbound 
calls and outbound calls to be in different contexts. Don't let your 
default context reach an outbound line. Your registered phones will be in 
a context that can call out which should be different from the default.

Also, make sure that your phones are registering with passwords (secret) 
that are different than the extension number. Makes it harder to guess.

The big thing to keep in mind dialplan wise is to never let an inbound 
call have a path to loop back outbound. The two of the biggest vectors for 
fraud will be allowing a non-authenticated sip call to get outbound over 
your trunks and to have weak credentials that can be cracked that will let 
someone else impersonate your phones.

And you can still wipe out most fraud by restricting the IP addresses you 
let in from the outside world. I prefer to have the most restrictive 
communications I can and then fix it if I discover that something doesn't 
work. Better to fail and fix than to permit and pay for it later. The 
providers I tend to like best not only give me what I need to restrict to 
their IP ranges, but also put in place restrictions on their end to only 
talk to my account from my external static IP address. That way someone 
could figure out my credentials, but if they can't spoof my ip address it 
still won't work. That is dependent on what the provider can do though.-- 
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Re: [asterisk-users] RES: RES: How to invoke a binary file from the dial plan?

[Kevin Larsen]

 I love this question, simply because it allows me to talk about one 
 of the neatest features I programmed into my system that barely 
 anyone knows exists. Plus it lines up pretty much exactly with what 
 you are trying to do. 
 
 We have our gate control system tied into our Asterisk phone system 
 so it is possible to dial a code on the phone and open the entrance 
 gate to let someone in after hours. Only problem is this happens so 
 rarely that no one (myself included) ever remembered the code. Thus 
 a search for a better way. 
 
 Now, when someone uses the gate phone to request entry, I change the
 caller ID on the display of the person who answers to read Press 9 
 to open gate. During the call, they can hit 9 at any time and the 
 gate will open for them. Up until they answer, the caller ID reads 
 Gate Phone, but when they answer, it changes to that text. 
 
 The part about opening the gate is the magic piece you want to look 
 into. Read up on applicationmap in features.conf. It's pretty simple
 and very effective. Here is what mine looks like. I am going to 
 replace my actual command with insert command here. 
 
 gate = 9,self/callee,System,insert command here ; Custom 
 application to open the gate. 
 
 This says that this feature is active in the 'gate' context of my 
 dialplan. The dialing pattern it is looking for is a 9. 'self' tells
 it to activate on the channel that dialed it and callee says that 
 the person receiving the call is the only one that can activate it 
 (otherwise the person at the gate phone could hit 9 to open it). I 
 am running the System dialplan application and passing it the 
 insert command here value. Everything after the ';' is a comment 
 as normal. The insert command here is equivalent to what you would
 put inside the '()' if it were in the dialplan (i.e. 'System(insert
 command here)'). 
 
 Pretty straightforward to get it working once you know what to look 
 for. Let me know if you want to know how I manipulate the Caller ID 
 upon answering the call to give the instructions to the callee on 
 how to open the gate/door. 

I just realized I said one piece wrong in this. 'gate' is not the context, 
it is the dynamic feature designator. I can illustrate this better by 
posting my front gate context.

[front_gate]
exten = number gate dials goes here,1,Set(__DYNAMIC_FEATURES=gate)
  same = n,Goto(frontgate_queue,${EXTEN},1)

-- 
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Re: [asterisk-users] Forward loop protection...

[Kevin Larsen]

 Deciding on the mailbox to use is problematic! The dialed-party may 
 be away for an extended period and wants voice mail handled by the 
 forwarded-to party.

And then you have the users who would work around this by sharing their 
voicemail passwords. Not quite as bad as sharing your computer log on 
credentials, but still, something I would like to avoid if possible.-- 
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Re: [asterisk-users] RES: RES: How to invoke a binary file from the dial plan?

[Kevin Larsen]

 Hi Kevin.
 
 Thank you very much for the hint! It worked very well!
 
 Your example ' exten = 1234,1,System(echo This is a test  /
 var/log/asterisk/test.txt) ' executes when the SIP client (my 
 softphone Jitsi) sends  a SIP INVITE to asterisk.  So, the softphone
 tries to establish a session with target 1234.
 
 Now, lets suppose my softphone rings and I answer a call. During the
 call, the caller asks me to execute a command (ex: to open a door or
 gate). In this case, what have I to program in dial plan to Asterisk
 execute System() again? Is it possible to execute a dial plan even 
 during an ongoing call?
 
 Finally, lets suppose I want to use my softphone to execute a dial 
 plan, even without establishing a call (no session with target 
 1234). For example, If I decide to open a dor or gate using my 
 softphone, without existing an ongoing call, what have I to program 
 in dial plan to Asterisk executes System(). Is this idea possible?
 
 Any hint will be very hepful!

I love this question, simply because it allows me to talk about one of the 
neatest features I programmed into my system that barely anyone knows 
exists. Plus it lines up pretty much exactly with what you are trying to 
do.

We have our gate control system tied into our Asterisk phone system so it 
is possible to dial a code on the phone and open the entrance gate to let 
someone in after hours. Only problem is this happens so rarely that no one 
(myself included) ever remembered the code. Thus a search for a better 
way.

Now, when someone uses the gate phone to request entry, I change the 
caller ID on the display of the person who answers to read Press 9 to 
open gate. During the call, they can hit 9 at any time and the gate will 
open for them. Up until they answer, the caller ID reads Gate Phone, but 
when they answer, it changes to that text.

The part about opening the gate is the magic piece you want to look into. 
Read up on applicationmap in features.conf. It's pretty simple and very 
effective. Here is what mine looks like. I am going to replace my actual 
command with insert command here.

gate = 9,self/callee,System,insert command here ; Custom application to 
open the gate.

This says that this feature is active in the 'gate' context of my 
dialplan. The dialing pattern it is looking for is a 9. 'self' tells it to 
activate on the channel that dialed it and callee says that the person 
receiving the call is the only one that can activate it (otherwise the 
person at the gate phone could hit 9 to open it). I am running the System 
dialplan application and passing it the insert command here value. 
Everything after the ';' is a comment as normal. The insert command here 
is equivalent to what you would put inside the '()' if it were in the 
dialplan (i.e. 'System(insert command here)').

Pretty straightforward to get it working once you know what to look for. 
Let me know if you want to know how I manipulate the Caller ID upon 
answering the call to give the instructions to the callee on how to open 
the gate/door.-- 
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Re: [asterisk-users] RES: RES: RES: How to invoke a binary file from the dial plan?

[Kevin Larsen]

 Hi Kevin.
 
 Thank you again for help me!
 
 In my case,  in the final application for smartphones or in a 
 softphone for PCs, there will be a button on the GUI and the user 
 will have just to touch it, and the door or gate will open. I mean, 
 during an ongoing call, the callee will see a button in the 
 interface of its SIP application. For example, we can use the lib of
 Linphone and implement a GUI over it, having a new button to open 
 doors and gates. So, the callee will not have to remember about 
 codes, because there will be a button in someplace to be touched.
 
 When the button be touched, during an ongoing call, the software 
 (SIP client) will sends a request to Asterisk executes the gate = 
 9,self/callee,System,insert command here , for example. So, it 
 will works like the user pressing number 9.
 
 I will take a look at applicationmap in features.conf to understand 
 what exactly can be done.
 
 But, let me ask you:
 This idea seems to be good to run during ongoing calls. What about 
 moments when there is no ongoing call? That is, can Asterisk execute
 a dial plan (maybe by means of some kind of SIP request received 
 from the SIP client) even without establishing a call?

The way I would probably approach what you want to do is that the button 
action state would be dependent on if you are in a call or not. If you are 
in a call, it sends whatever DTMF digits you want to use for this feature. 
If you are not in a call, it could dial an extension whose purpose is to 
do the same thing. 

I have an outside number that when dialed checks that your caller id 
number is in an approved list and if it is, sends the gate open signal. 
This is the same gate open signal that the feature code uses (the call to 
System()), it is just reached by making a sip call. Nothing says a call 
has to connect two phones together. You can answer the call inside of 
Asterisk and do stuff based on what number you called or what digits the 
caller enters with their keypads. Lot's of opportunity to make the system 
do exactly what you want.-- 
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Re: [asterisk-users] Forward loop protection...

[Kevin Larsen]

  Ia had a server overload today because someone did a call forward 
 to their own extension.  To do a call forward I write a key called CFWD 
 with the extensión number and number to dial .  The main script tests if 

 the key/value exists and dials the number stored in the database.  What 
 is an easy way to prevent dumb people from creating a loop?

Right after you have read the number to call forward to, compare it to the 
number you are call forwarding from. If it matches, play the user an error 
message and have them try again.

And no matter what you do, the dumb people will come up with more creative 
ways to tank your phone system. A large amount of my dialplan code is 
taking into account the stupid things they have done and handling it 
properly if they do it again. I swear, if you could harness their 
creativity for good you could solve the world's problems 10 times over.-- 
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Re: [asterisk-users] How to invoke a binary file from the dial plan?

[Kevin Larsen]

 Hi everyone.
 
 I'm new with Asterisk and I have to create a dial plan that will 
 invoke a binary code. That is, asterisk will execute a program in 
 the same machine. How to do it?
 
 Let me explain what I have to do:
 
 In the project that I am currently working, there is smartphones, 
 SIP servers and doors/gates to be unlocked remotely. When the user 
 executes an application on his/her phone, it will presents a button 
 to unlock a remote gate or door.
 By  pressing such button, the application will send a SIP INVITE to 
 the SIP server (Asterisk). In this moment, a existing dial plan 
 should call an executable hosted in the current machine. In this 
 case I need to know how to program my extensions.conf to let 
 Asterisk invoke another software to me.
 The another software is the one responsible for unlocking a gate or 
door.
 
 So, how to codify my extensions.conf in order to make Asterisk 
 invoke another software?
 Is another better way (idea) to implement my project using Asterisk 
 and SIP? If so, comment, please!
 
 Any hint will be very helpful!

Look into the System() dialplan application. It will execute a command on 
the system for you. Be aware that it will execute it as the user your 
Asterisk instance is running as, so permissions can sometimes be a bit 
finicky to get correct. I do something similar to pop my gate open. It is 
using nc to make a connection to the device, but same general idea as what 
you are doing.-- 
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Re: [asterisk-users] RES: How to invoke a binary file from the dial plan?

[Kevin Larsen]

 Ok. Thanks for the hint.
 
 But, what exactly is a System() dialplan application? Is it a kind
 of command that i can call in dial plan? 
 
 I will look for System() related to dial plans.

From the Asterisk CLI type:
core show application System

It will print out the syntax for the command. One of the easier dialplan 
applications. 

exten = 1234,1,System(echo This is a test  
/var/log/asterisk/test.txt)

That line would use the Linux echo command to place the text This is a 
test into a file named test.txt located in the /var/log/asterisk 
directory.-- 
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Re: [asterisk-users] Forward loop protection...

[Kevin Larsen]

 The loop checking is a bit more challenging than that. If Bob 
 forwards to Fred and Fred forwards to Sue, all is well when Bob and 
 Fred head out for a beer. A little later, we’re in deep doo-do0 when
 Sue forwards to Bob. 

 Could this possibly mean that any person who has CF set should never
 be available as CF Destination. Simple db entry/check can have this 
done.

That just goes to show that the problem can get complex pretty quickly. 
Using the original example above, it might be that you want to allow the 
Bob to Fred to Sue forwards, but only stop it if the Sue to Bob link is 
established, thus creating the loop. I wonder if you could do some kind of 
recursive check where you follow each forward and if you ever come back 
around to a number you have already checked you know there is a loop.

To reuse the example above, on the creation of the Bob to Fred forward, 
the database is checked to see if Fred has any forwards. He doesn't, so is 
at the end of the forwarding chain. Now Fred forwards to Sue. Again, she 
is at the end of the chain, so it is allowed. When Sue goes to forward to 
Bob, the check shows that Bob has a forward. Not a problem, but we create 
a temporary list that has Sue's number in it. Then we check the next stage 
of forwarding. Bob forwards to Fred. Fred's is checked against our 
temporary list and doesn't match, so we are still good. Bob's number is 
now added to the temporary list and we check the forward Fred has in 
place. Fred forward's to Sue. We check Sue's number against the temporary 
list and it does exist. Thus we have a loop detected and the forward can 
now be denied.

I am guessing with the recursion involved you might want to do the check 
outside of Asterisk and pass the result back in. I will also state that I 
have not had to do this deep checking in the past, so these are just some 
initial thoughts on how I would start approaching the problem. Of course, 
this also assumes that Bob, Fred, and Sue are all on the same phone 
system. If you don't have a shared database to look at, the problem just 
got harder indeed.
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Re: [asterisk-users] Signaling incoming call

[Kevin Larsen]

 Hi Kevin!
 
 Thanks! It works!
 I can set the name of the line with CALLERID(name) and see the caller 
number,
 too.
 And, it the number is in the address book, I see the name, too.
 
 Perfect!

Glad it worked for you. I usually leave the number untouched, but will 
manipulate the name to suite what I want. I have mulitple call queues, so 
for instance, for my helpdesk lines, I will do something like transform 
Name to HD:Name so that the person being called knows that the caller 
dialed the help desk number rather than their direct number. On people who 
work multiple queues, it is very handy so they can see at a glance what 
queue the caller is reaching.-- 
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Re: [asterisk-users] Signaling incoming call

[Kevin Larsen]

 Hi Steve!
 
 Thank you very much!
 It seems to run!
 
 I wrote that:
 
 exten = _0049351333,n,Set(__ALERT_INFO=Bellcore-r3)
 exten = _0049351333,n,SIPAddHeader(Alert-Info:
http://www.notused.com
 \;info=alert-external\;x-line-id=0)
 
 and the phone rings with another melody.
 Very curious is, that if I don't write BOTH lines, it does not run...
 
 And, unfortunately, I just have two melody: the normal and this one, 
but it
 is better than nothing!
 Now, if it will be possible to add a text on the display, it will be 
perfect,
 but I didn't found any option for that...

Look into Set(CALLERID(name)) and Set(CALLERID(num)) to manipulate the 
caller id name and number that show up on the phone.-- 
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Re: [asterisk-users] Peer is UNREACHABLE

[Kevin Larsen]

  What kind of phone are we talking about, both yours that works and 
your 
  wife's that does not?
 
 Right!
 
  Can you ping the unreachable phone and does it respond to a ping?
 
 I can ping both phones from the VM
 
  Many phones will have a network test function built in to them to help 
you 
  determine if the phone is properly connected to the network.
 
 Unfortunately not that...
 I tried with Twinkle from my PC, using the same account of my wife
 (configured IDENTICALLY to my account, just another username). It don't
 work...
 I presume, I configured something wrong in Asterisk...
 
  Do you see anything in the asterisk logs or the logs of the phone 
itself 
  (providing the phone puts logs somewhere) that indicate a failure to 
  register or to resolve the ip address of the asterisk server?
 
 Unfortunately not... Just UNREACHABLE...

Can you post the Manufacturer and Model of your phones (both of them if 
they are different)? That will help us look up what diagnostics/log files 
there might be on the phones.

Does the Twinkle software on the PC show any error messages?

If you watch the CLI in asterisk, does anything go by in there regarding a 
failed registration? If I get one of my phones programmed with an 
incorrect username/secret, it will try to register with the server, but 
can't. Those failed registrations do show up in the CLI.

Double check that you are not mistyping the credentials somewhere. If you 
do post the relevant parts of your config in here, you might want to 
obscure the secret.-- 
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Re: [asterisk-users] Peer is UNREACHABLE

[Kevin Larsen]

 I have a problem and I hope someone can help me...
 I configured an Asterisk on a VM to serve more accounts and act as a 
proxy to
 other SIP-providers.
 
 The first account running on my phone works without any problem.
 A second account, running on the phone of my wife, is always 
UNREACHABLE.
 I can just see in the log:
 
 [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer
 '004935111' is now UNREACHABLE!  Last qualify: 0
 
 In the CLI I can see:
 
 Name/username  HostDyn Nat ACL Port Status  
 004935111/0049351  192.168.200.11   D  5060 UNREACHABLE 
 004935122/0049351  192.168.200.10   D  5060 OK (17 
ms) 
 004935133  (Unspecified)D  5060 UNKNOWN  
 
 1234   (Unspecified)D  5060 UNKNOWN  
 
 messagenet/1234567890  212.97.59.765061 Unmonitored 
 pbxanika/004935172.16.34.132   5060 Unmonitored 
 pbxfax/0049351333  172.16.34.132   5060 Unmonitored 
 pbxluca/0049351222 172.16.34.132   5060 Unmonitored 
 8 sip peers [Monitored: 1 online, 3 offline Unmonitored: 4 online, 0 
offline]
 
 Asterisk connects to another Test-VM with AsteriskNOW and to the italian
 provider Messagenet.
 
 Can someone suggest me, what can I do?
 I can send the configuration file, if they are needed.
 

What kind of phone are we talking about, both yours that works and your 
wife's that does not?

Can you ping the unreachable phone and does it respond to a ping?

Many phones will have a network test function built in to them to help you 
determine if the phone is properly connected to the network.

Do you see anything in the asterisk logs or the logs of the phone itself 
(providing the phone puts logs somewhere) that indicate a failure to 
register or to resolve the ip address of the asterisk server?-- 
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Re: [asterisk-users] Peer is UNREACHABLE

[Kevin Larsen]

 Darryl Moore dar...@moores.ca schrieb:
 
  I'd start by turning on sip debugging in asterisk
   sip set debug ip [your_phone_ip]
 
 Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d@172.
 16.34.133' Method: OPTIONS
 Reliably Transmitting (no NAT) to 192.168.200.11:5060:
 OPTIONS sip:0049351222@192.168.200.11:5060 SIP/2.0
 Via: SIP/2.0/UDP 172.16.34.133:5060;branch=z9hG4bK13db26f5;rport
 Max-Forwards: 70
 From: asterisk sip:asterisk@172.16.34.133;tag=as1215345d
 To: sip:0049351222@192.168.200.11:5060
 Contact: sip:asterisk@172.16.34.133
 Call-ID: 78f3a0d0145f3dfa630a5e7c506142d6@172.16.34.133
 CSeq: 102 OPTIONS
 User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4
 Date: Thu, 28 May 2015 20:39:02 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 Content-Length: 0
 
 repeated in loop...
 Help that?
 
 192.168.200.11 is the IP of the phone of my wife, and 172.16.34.133 
 the IP of the Asterisk server.
 

The phone you gave your wife is really old. Are you sure it supports SIP 
OPTIONS? Can you make a call in or out to it? If you can, it is more 
likely that it just doesn't support that and you can't use a qualify 
statement.-- 
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Re: [asterisk-users] Peer is UNREACHABLE

[Kevin Larsen]

 No, I'm not sure.
 And no, I can't make any call, right now... At least, not connected to 
my
 Asterisk...
 If I connect it to the other VM with AsteriskNOW I can call my Twinkle, 
but
 NOT my phone connected on my Asterisk, using the proxy.
 I can see that in the log:
 
 [May 28 22:49:51] WARNING[4135]: chan_sip.c:12800 check_auth: username
 mismatch, have 1234, digest has luca
 [May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite:
 Failed to authenticate device Test1 
sip:1234@172.16.34.132;tag=as6dd12e05
 

I know from your previous email that you are new to Asterisk. Have you 
created a dialplan that would allow you to call from one extension to 
another without going through your phone company? That is to say, call 
from your phone through Asterisk to your wife's phone?

You have two parts that you need to have in place for the basics to work. 
You need your sip.conf in order to tell asterisk what devices and phone 
trunks you have and you need extensions.conf to tell Asterisk how to route 
calls. Since you are new to this, you can start by getting the two phones 
to both register (sounds like one of them is and one probably is not). 
Then you get to where you can dial from one phone to the other and vice 
versa. From there you can add in the telephone company lines and the 
ability to dial in and out to the world.

I am still curious why you have both an Asterisk setup and an AsteriskNow 
setup? Is that just to play around with? At the end of the day you should 
just need one or the other.-- 
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Re: [asterisk-users] Asterisk as Proxy and more device for a number

[Kevin Larsen]

 I'm very new in Asterisk and VoIP, and of course I have a problem... :)
 
 Well, my problem is, that Deutsche Telekom wants me to change my ISDN 
 to VoIP... :(
 I must do that, since I have no alternative.
 
 Well, I have now two VoIP-phones (Thomson ST2022 and KE1020A). I can 
 configure my two numbers by Deutsche Telekom and I got now an extra 
 number from Messagenet.it.
 
 Now the problems:
 1) It seems that I can't configure my ST2022 to have two profiles and 
 both are running on different servers
 2) I want that when a number will be called, both phones rings
 
 I think, I need an Asterisk-Server between my phones and the 
 VoIP-Provider, isn't it?
 
 Well, now the questions: am I right? Should I install an Asterisk on 
 my PC to do that?
 And of course: how can I do that? How can I set up Asterisk to serve 
 as proxy for these three numbers and send the calls to a number to 
 both phones?
 
 Unfortunately, I didn't found any HowTo for my problems...
 

If you want to go the Asterisk from scratch route, you would do well to 
pick up a book on the subject. Since you seem comfortable with English, 
Asterisk: The Definitive Guide is a good place to start. This will teach 
you how to build an Asterisk system from the ground up. Depending on what 
you want to do, this may also be overkill.

There are Asterisk distributions that already come with a GUI front end 
that could make this all a lot easier to set up. AsteriskNow (includes 
Asterisk and FreePBX Gui) is a good choice as would be Elastix (Asterisk + 
FreePBX GUI + the Elastix GUI). These are often much easier to set up for 
the Asterisk newbie. Either of those should be able to easily handle what 
you want to do.

Even if you do go the route of a pre-made distribution with a GUI, the 
Asterisk book is still useful to have. It really gives great insight into 
the software and will help if you ever have to troubleshoot from the 
command line.-- 
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Re: [asterisk-users] Asterisk as Proxy and more device for a number

[Kevin Larsen]

 Maybe I got it...
 I installed an asterisk on a VM with Ubuntu 10.04 and I got it 
connecting to
 another Test-VM with AsteriskNOW and with an italian VoIP-provider.
 The very difficult was to understand, that my phone just can manage ONE
 profile at time, so I had to configure Asterisk to receive all calls 
from the
 different providers an send them ONE profile (on my phone).
 
 Next step is to configure Asterisk for the other phone (for my wife) and
 having all calls of her number forwarded to my phone and her phone.
 
 Next step again is to manage outgoing calls going to the right provider.
 
 Then it would be nice if I can forward calls from a phone to the other.
 
 Last but not least, I need to use HylaFAX on an account on Asterisk.
 I had many problems with T38Modem, so I'll try with IAXModem, maybe I'll 
got
 it...

Glad you have it working. You should only need one Asterisk server to do 
what you want unless you just want to have one with the GUI and one for 
testing purposes. I would recommend starting with something newer than 
Ubuntu 10.04 as it is pretty much at its end of life. 14.04 would be a 
better choice at this point.

Regardless of how you end up directing your incoming calls, that KE1020A 
phone is pretty old and it might be worthwhile to see about upgrading it 
to something newer. The Thomson ST2022 you have does seem to have the 
capability to have two lines on it. Haven't used one before, so hard to 
say how good it handles that. Whatever you do, though, having two 
identical phones will be helpful to you (and your wife) as you won't have 
to try to remember how each phone works and troubleshooting problems is 
easier if you can look at a phone that is working of the same model.

There are a couple of ways you can approach directing your calls to the 
right outgoing provider. One would be to have two separate lines on your 
phone and just pick which one you want to use that will direct all calls 
to the right provider. If your calls follow a pattern (i.e. calls to this 
country go to this provider and calls to that country to to the other 
provider), you can have Asterisk recognize the pattern and automatically 
direct the calls for you. This is nice as others won't have to remember 
which line to use.

Asterisk has built in forwarding capabilities by dialing the right feature 
code during a call to initiate a forward to another extension. Many phones 
also have this feature built in. I use Polycom phones and can transfer 
calls just by hitting the transfer button and dialing who I want to 
transfer to.

I have used the HylaFax/IAXModem solution with a client and it worked 
fairly well. I will warn you that faxes over VoIP connections are 
inherently worse than over a regular phone line. They can be made to be 
almost as good or they can just be horrible, but either way, faxing is no 
fun, especially considering that the problems can be caused before the fax 
ever reaches your system. Hopefully your provider supports T.38 properly, 
in which case faxing will be much nicer.-- 
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Re: [asterisk-users] Phone provisioning template Snoms

[Kevin Larsen]

  I am looking for a phone provisioning template for Snom phones, 
 Yealinks and Polycoms. I am always doing deployments of many phones 
 and usually configure each phone one by one for each installation. 
 Any help will be highly appreciated
 
 There’s some excellent documentation about provisioning on the Snom 
Wiki:
 
 http://wiki.snom.com/Category:Auto_Provisioning:Configuration_Files
 
 You can set the phones (via DHCP options) a firmware url on a web 
 server under your control, grab their MAC addresses, then deliver 
 them custom config settings as required.
 
 Easiest way to start is to copy the config file (via the web 
 interface) from a phone with factory default settings, then just 
 change the settings you need to change, and write something in your 
 scripting language of choice (PHP, Perl, Python, etc.) to just send 
 those settings to the phone dependent on MAC address. Don’t send 
 *every* available config setting to the phone - only the changes 
 from default you need to make.
 
 I suspect the same can be done with Yealink and Polycom phones - 
 I’ve not used those so can’t really comment. I have a similar system
 which seems to work for Sipura/Linksys/Cisco phones, though most of 
 my new deployments are exclusively Snom.
 

I use the Polycom phones and there are any number of ways you can automate 
deployments of them. The templates you want to start with can be found on 
the Polycom website here: 
http://support.polycom.com/PolycomService/support/us/support/voice/index.html

When you download the firmware (UC software release), you get the 
templates you want included in the download. You can use FTP, TFTP, and 
HTTP that I know of to provision the phones. I use HTTP and have some 
custom php scripts that I wrote that create my own templates on the fly 
for a phone based on its mac address. You can use a combination of static 
templates for things that are system wide and dynamic templates for the 
things that are specific to each phone. You can also just create a static 
template for each phone. It really depends on how in depth you want to go. 
It does make provisioning much easier though.
-- 
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Re: [asterisk-users] Multicast to polycom from asterisk

[Kevin Larsen]

 I am using 11.17.0 -  and MulticastRTP. Doesnt seem to work with 
 polycom phones as other devices receive my multicast just fine.
 
 Is there something special to do to get multicast working with polycom 
phones?
 (other than enable multicast on the actual phone).

Didn't see if anyone had answered you or not on this, but Polycom uses 
their own form of MulticastRTP. It doesn't work with Asterisk's multicast 
setup. There is a company that makes a loud ringer/pager unit that can 
also be used to take in a sip call and multicast out to the Polycom 
phones. I haven't tried it myself as I just use the loud ringer 
capability, but it does appear that it would be a workable solution. I 
hesitate to promote the name here since this is non-commercial discussion, 
but let me know if you want to hear the actual product.-- 
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Re: [asterisk-users] Multicast to polycom from asterisk

[Kevin Larsen]

  I hesitate to promote the name here since this is non-commercial 
  discussion...
 
  but Polycom...
 
  Polycom phones...
 
 If mentioning Polycom is OK, I think mentioning a possible commercial 
 solution is OK.

In that case, the product in question is the Algo 8180 SIP Audio Alerter. 
I will state that I have not used this particular functionality, but it is 
mentioned in the users guide, so in theory you could use it as a bridge 
between Asterisk and mulitcast on the Polycom phones. YMMV.-- 
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Re: [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 03/25/2015 01:38:26 PM:
 I'm looking at enabling autopause on one of my queues where my queue
 members are bad about leaving their desks without pausing.

 The problem I see is that when the queue pauses an Member it doesn't
 jump into the dialplan to do so which means my handy device state 
 and asterisk database driven Light for the Member showing their 
 paused status won't update.

 My idea for solving this problem is to check the status of my Member
 in the queue before I send the calls into it and toggle on the 
 Members Paused light at that point in time if they are paused.

 Sadly I don't see a way to determine if my Staff are paused or not 
 from the dialplan, There doesn't appear to be a function to retrieve
 the status of the members in the queue.

 Does the list have any suggestions?

First, let me say I feel dirty for even posting this. It is probably far 
from ideal, but it does get the job done. I had the same issue. Also, I am 
using Asterisk 11. I just looked and it doesn't appear that the 
QUEUE_MEMBER function supports the paused option in 1.8. To be honest, I 
am not sure if there is a good replacement for what I have done below in 
the 1.8 series.

[sub_autopause_status]
exten = s,1,NoOp(Checking for autopaused members for ${arg1} queue)
  same = n,Set(MEMBERS=${QUEUE_MEMBER_LIST(${arg1})})
  same = n,Set(i=1)
  same = n,Set(max=${FIELDQTY(MEMBERS,,)})

  same = n,While($[${i} = ${max}])
  same = n,Set(MEMBER=${CUT(MEMBERS,\,,${i})})
  same = n,Set(STATUS=${QUEUE_MEMBER(${arg1},paused,${MEMBER})})
  same = n,Set(MEMBER_EXT=${CUT(MEMBER,\/,2)})
  same = n,ExecIf($[${STATUS} = 0]?System(echo IN  
/var/spool/asterisk/status/agent-${MEMBER_EXT}-status.txt))
  same = n,ExecIf($[${STATUS} = 1]?System(echo PAU  
/var/spool/asterisk/status/agent-${MEMBER_EXT}-status.txt))
  same = n,NoOp(${MEMBER}: ${STATUS})
  same = n,Set(i=$[${i} + 1])
  same = n,EndWhile()

  same = n,Return()


So, as an explanation, I have multiple queues and agents who autopause. I 
show their status on their phones, hence the System(echo...) commands to 
the /var/spool/asterisk/status directory. Those files are used to generate 
a simple web page that is shown on their phones that lets them see their 
status. You should be able to adapt that to what you do.

Basically, you pass the queue name into the subroutine as arg1. The 
subroutine gets a list of every person logged into that queue and then 
loops through checking the status of each person using the QUEUE_MEMBER 
function.

It isn't elegant and if you have a lot of queues/queue members to check, 
it will constitute a lot of looping, but it does work. Like you, I would 
like to have a way to check the pause status of a member easier. If the 
queue application could call a subroutine with it autopaused someone, that 
would actually make an elegant solution, but for now, this was the way I 
could see to do it.

You could maybe call a script that would parse the queue_log file looking 
for an agents status and pass that back into the dialplan.-- 
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Re: [asterisk-users] [OT] switches

[Kevin Larsen]

 so how does a client pc find the server if there's no NAT?  by IP 
 address?? That makes no sense, to me, if the switch isn't assigning 
 addresses.

Switches have a MAC table that keeps track of which MAC addresses are on 
which ports. That's how they decide where to route packets.

http://en.wikipedia.org/wiki/CAM_Table
http://en.wikipedia.org/wiki/OSI_model-- 
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Re: [asterisk-users] System() command refuses to execute bash script

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 03/02/2015 08:27:07 AM:

 From: Stefan Viljoen viljo...@verishare.co.za
 To: asterisk-users@lists.digium.com, 
 Date: 03/02/2015 08:27 AM
 Subject: [asterisk-users] System() command refuses to execute bash 
script

 How can I use System to run a bash script?

Just to rule out some weird permissions issue, try to write the file to 
some directory that has full read/write permissions to everyone (eg 777). 
If the file can be written to that directory you probably have a 
permissions issue still. I run my asterisk under the asterisk user and 
have it kick of scripts that write to a folder on the system all the time. 
The folder has full permissions for the Asterisk user.

Give it a shot and see what you get.-- 
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Re: [asterisk-users] BlindXfer Sensitivity

[Kevin Larsen]

 Hi Guys
 
 We have a client running on a polycom vvx400 IP phone on our 
 asterisk 1.8.18 system
 
 The issue we have is the switchboard lady uses ## to transfer calls 
 but sometimes it just does not work and just plays the DTMF tone to 
 the calling party.
 
 Is there any way to adjust the sensitivity of the blindxfer feature?
 
 The polycom Transfer button is useless  as there is a big delay 
 until it apprears
 
 I would greatly appreciate any advice

It seems weird that this would be some kind of sensitivity to the DTMF 
tones. The first thing I would look for is on a call that she cannot blind 
transfer, check how the Dial command was used to reach her. Does it have 
the proper use of the tT options (depending on whether she called them or 
they called her)? I would almost bet there is a call path that occurs 
which doesn't have the proper options set to allow the transfer.-- 
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Re: [asterisk-users] Is Asterisk a Linux only system?

[Kevin Larsen]

 I know that it runs on other systems but do other ports get the same
 attention?  I have been running it on a NetBSD server for about a year
 now and while it mostly works it just crashes from time to time with no
 explanation or core dump.
 
 I have improved the situation by expanding my intrusion detection but
 it still stops every few days or so.  I have a cron job that tests for
 it and restarts it when necessary.
 
 Anyone else have experience on non-Linux systems?

I have not run it on a non-Linux system, but for monitoring and restarting 
it when it fails, you might look into Monit. That might be more efficient 
than waiting for a cron job to check it and restart.-- 
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Re: [asterisk-users] JITTERBUFFER function

[Kevin Larsen]

 WTF is a jitterbuffer?

http://lmgtfy.com/?q=jitterbuffer
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Re: [asterisk-users] Dialing from phonebook, and hiding the dialed number from the user.

[Kevin Larsen]

 Hi,
 
 does anyone have a recommendation for a SIP phone, which
 allows dialing from a phonebook, and hiding the dialed number
 from the end users? Also from the call history of course.
 
 It seems Mitel can do this, and I have a use case where this is
 a requirement.

I don't know about a phone that can do that, but I can give you another 
possibility that might be an acceptable substitute.

You could alias the numbers in the phone so that in Asterisk they do 
something different. In the phonebook you would have something like: Bob 
Smith: 1000. Then in Asterisk, you have as part of your dialplan that 1000 
would dial Bob Smith's real number. The user of the phone would only ever 
see the number 1000 associated with Bob Smith. The history would still be 
there in the phone, but again, it would just show 1000 as well.

How far you take this would depend somewhat on how often the underlying 
numbers change. You could hard code the numbers in your Asterisk dialplan 
or you could plug them into a database so that they are easier to change 
in the future. Would that work for what you need?-- 
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Re: [asterisk-users] Polycom SoundStation 6000 Dropping Registration

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 01/23/2015 10:24:24 AM:
 Hello,
 
 I'm having a problem with a few Polycom SoundStation 6000s. 
 Everything works fine, but they drop registration to asterisk after 
 about maybe 30 minutes – the phone does not re-try to register and 
 if you try to dial out on the phone it says “URI Dialing is Disabled”
 
 Has anyone else had this issue? I'm running asterisk 11.7.0.

We run a variety of 5000, 6000, and 7000 series Soundstations running 
Asterisk 11.6.0 and the phones are at 4.0.3.7562. We do not see these 
registration issues.
-- 
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Re: [asterisk-users] About voip gateway

[Kevin Larsen]

 I want to create a voip service, I do not know much about it, but 
 the first thing I want to know if more than one client can make a 
 call at the same time through internet to the PSTN, and what gateway
 should I use for this.
 
I think the first recommendation any of us will have is to research all 
you can as there are a lot of mistakes to be made in the telephony world 
and some of them can be expensive and/or dangerous. The kinds of questions 
you are asking are not bad to ask, but they do place you squarely at a 
beginner level.

It is hard to answer your questions without having further information. 
What are you trying to accomplish with this system? Do you need to carry 
more than one call? What types of phone service are available where this 
will be installed?

For example, a single POTS line will allow you one call in or out of the 
PSTN. This is not a limitation of Asterisk, this is a limitation on how 
POTS lines work. A PRI style connection (E1 or T1 depending on location) 
will allow many more (over 20 calls at once). A SIP trunk is only limited 
by the number of lines your trunking provider allows and the bandwidth of 
your internet connection. The gateway you would want to use will depend 
entirely on what type of connection to the PSTN you are using. A lot of 
manufacturers make hardware compatible with Asterisk for physical 
connections to the PSTN and a SIP trunk just requires an internet 
connection of sufficiently high bandwidth, low latency and a reasonably 
stable path to the SIP provider.

Without knowing more about what you are aiming to do, it is hard for 
anyone to give you any specific help. You were earlier asked for a 
specific example of what you wish to accomplish. Please provide that and 
you will get more people responding.
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Re: [asterisk-users] OT - Is T.38 possible on SPA8800 FXO port ?

[Kevin Larsen]

 I know all this.
 
 My question came from the fact that as strange as it may seem, SPA3102
 and similar products do not offer the SIP features depending on
 terminating/originating port.
 More precisely, when a SIP fax call comes in through an FXS port, it
 triggers T.38 while it doesn't trigger T.38 when an FXO port is used
 instead.

So, if i understand the question correctly, you have:
 Asterisk SIP- SPA3102 FXS - Analog fax machine
and the PBX to SPA3102 communications are T.38 before converting to analog 
to go to the fax machine. Then in the other situation you would have:
 Analog line - SPA3102 FXO port - Asterisk SIP
and the SPA3102 to Asterisk communication isn't doing T.38?

If that is the case, the only thing I can think of is that maybe they were 
not thinking that many people would want to do the second situation with a 
low end device? I imagine the main use case it to keep the old analog 
device around after switching to SIP delivery. Probably didn't expect to 
see analog delivery that gets translated to a sip fax endpoint. 

Though I do agree that once you have the transcoding option, I would think 
it would be trivial to apply it to both ports.-- 
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Re: [asterisk-users] E1 - Cisco - Asterisk and vice verso

[Kevin Larsen]

  Hi,
 
  Change my Dynastar E1 gateway to Cisco with E1 module, but can't make
  easiest dialplan. All my routing i made on asterisk, so i need that 
cisco
  all calls from E1 send via sip to Asterisk and all calls came from 
Asterisk
  by sip send to E1. From E1 to Asterisk already work, but can't 
understand
  how send all from Asterisk SIP to E1 ?
 
  Can you help ?
 
 If you are taking SIP calls into Asterisk and want to send them out
 E1, you need an Asterisk-compatible E1 board, such as:

I think, though I am not for sure, that he is asking how to route a call 
from Asterisk to the Cisco device that is acting as the gateway (i.e. the 
E1 is connected to the Cisco and will only be speaking SIP to Asterisk). 

We had a similar setup before we replaced the Cisco with Digium Gateways. 
Basically we just set up a peer in asterisk with the IP address of the 
Cisco and routed external calls to it. The Cisco then actually placed the 
calls to the PSTN. Pretty painless, though sometimes a bit confusing to 
remember that if there were issues, you might have to check dialing rules 
in both Asterisk and in the Cisco. -- 
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Re: [asterisk-users] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)

[Kevin Larsen]

 From: Paul Albrecht palbre...@glccom.com

 Seems like now is as good a time as any to raise these issues, in 
 fact, sooner is better than later because once developers start down
 a path it’s very difficult to get them change their minds no matter 
 how much sense it makes. The fact that developers are even 
 considering taking away user functionality like the dial plan is in 
 of itself a very serious problem because it demonstrates they don’t 
 see Asterisk from the user perspective.

 Don’t object to extending the Asterisk user interface or changing 
 Asterisk internals. Do object to is taking away functionality that 
 users expect, are familiar with, and has made the Asterisk project 
successful.

 Then your experience is atypical. Asterisk has been unstable for 
 several years as developers have continually shoveled new features 
 into the code base over several releases. That’s not necessary 
 objectionable, it’s even to be expected; however, at some point 
 developers need to turn their attention to less glamorous less 
 exciting things like stability and performance.

I don't think anyone is objecting to you bringing this up, as it has been 
mentioned at the dev con. Perhaps it is just that the tone doesn't come 
across properly in an email, but you are coming across as confrontational 
and alarmist and it seems to be setting people on edge. Matt has already 
chimed in that he doesn't see how it would be possible to deprecate the 
dial plan at this time and even if it were possible, the process would 
take on the order of years, giving you plenty of time to enact any 
contingency plans you might need. Scott G. from Digium even posited that 
if it were to be removed from the core, it would likely end up as a 
loadable module so that it wouldn't burden those who don't need it and 
could be loaded for those who do.

These developers do not exist in a vacuum, nor do they have total control 
over where Asterisk goes. Influence, sure, but there is still a corporate 
structure out there that finds it necessary to be customer oriented. They 
would have to be monumentally stupid (something which I haven't seen 
previous evidence of) to kill off the dial plan without providing a path 
forward for those who depend on it. Furthermore, even if they did pull a 
stunt so bad as to alienate half their users, the open source code would 
be forked so fast as to make your head spin or people would migrate to 
other similar packages (Freeswitch comes to mind). Digium sells their own 
PBX hardware that I am sure depends on these technologies that you are 
afraid will go away. They have direct skin in this game too.

I would be interested to know just how atypical my experience is. I have 
found that on my 1.6 systems I would have random crashes over time. After 
upgrading over multiple sites, my 11.x systems have been rock solid for 
the most part. I did have a case where I did a store and forward of a fax 
that if I tried to forward the fax and it had no file to forward would 
cause a crash, but other than that, I haven't seen any problems in normal 
day to day usage. I always thought that the general consensus was that the 
11.x series was quite a bit more stable than the older versions.

Kevin Larsen
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Re: [asterisk-users] Asterisk- VoIP Phones ring from Ext. 101 but that Ext. does not exist in extensions.conf

[Kevin Larsen]

 Hello, 
 a user outside the office regularly gets a call from ext. 101 but 
 that extension does not exist in my extensions.conf. when the user 
 pickup the phone no one answers. Any Idea how to fix this issue? 
 that user uses Polycom SP 450, 

First thing to look at is at the time the user receives the call, do you 
show anything in your Asterisk CLI? I would make sure that the call is 
actually originating from your system and track back from there.-- 
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Re: [asterisk-users] Record ANSWERED call

[Kevin Larsen]

 The problem is it records all incoming calls include those with the 
 disposition of NO ANSWER, FAILED, BUSY, UNKNOWN.. For example the NO 
 ANSWER call will leave a 44byte wav file in my ${RECDIR}
 
 How can I record only the calls with the disposition of ANSWERED?
 
 May be I should run a cronjob to clean up the 44byte file after it's 
 been created? Is there a better way?
 

I would probably add a line in my hangup (h) extension that does an execif 
to delete the file if the disposition is not equal to ANSWERED.-- 
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Re: [asterisk-users] Call Flow Documentation Tools

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 09/12/2014 09:07:36 AM:

 I have been researching software for documenting pbx call flow paths
 and I was just wondering if anyone out there is using anything they 
 have found particularly useful or cool.
 
 I am looking for something preferably visual that the average end 
 user can follow. So far the best thing I have come up with is making
 a diagram with a decision tree in visio but its very time consuming 
 to build this by hand for every customer. We would like to be able 
 to provide every customer a diagram so they can easily understand 
 the path that a call takes, what conditions are checked and what 
 actions are taken based on those conditions.
 
 A large portion of my asterisk installs are for non profit or 
 charitable organizations so while I'm not completely fixed on a free
 solution, if it isn't free the cost needs to be relatively low or at
 least be a multi-tenant solution that  could at least be used for 
 multiple customers.
 
 For most of our installs we manage everything via CLI, but for a few
 orgs with tech savvy people we have been able to setup freepbx for 
 them and let them make simple changes. I was thinking with freepbx 
 maybe there could even be a module that takes the freepbx 
 configuration and generates visuals based on reading the 
 configuration, this would be really slick although not a complete 
 answer as we have many installs that do not have freepx.
 
 Anyway, just wanted to get some input from others and put my ideas 
 so far out there, if you have any recommendations or experiences to 
 share feel free to reply on or off-list.

I see where you are going with this, but haven't seen anything that can 
analyze a dial plan and generate a flow document. The closest I know of is 
something like Apstel's Visual Dialplan (http://www.apstel.com) which is 
meant to go in reverse (visually create the dialplan and it creates the 
Asterisk code). I have never used it so can't comment on if it can print 
out the information in the form you are wanting.

I can see that what you want would be difficult to create, but would be 
very handy. I am interested to see if anyone else knows of such software.-- 
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Re: [asterisk-users] Special functionality for Secretary/Boss

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 09/04/2014 11:57:40 AM:
  We are currently migrating from a Nortel pbx to Asterisk and we 
 have been able to convert most of the functions that people are used to 
 but there is one I have no clear idea how to do.  The scenario is:
 
  Boss calls secretary from outside the office to get connected to 
 another outside destination.  The secretary dials the destination and 
 then trasfers call to the boss.  When boss finishes with that person 
 they want to send the call back to the secretary in order to make 
 another connection or simply to talk to the secretary.
 
  The first part is not a problem, but after the boss finishes his 
 call how can we send the call back to the secretary?  I was thinking of 
 using a conference room but how would the secretary know when the boss 
 has finished?  Anyone know how to handle this scenario?

I haven't tested this, but my initial thought would be to create a special 
context or extension that the secretary could route through when doing the 
call transfer. The Dial application could be called with the 'g' option to 
continue the dialplan at the next priority when the call hangs up. 
Something like a normal call transfer would just dial the number as 
normal, but for the special transfer, you could prepend the dialed number 
with a #.

For example (using a local US dialstring, change to fit your needs):

; This is a normal external call.
exten = _NXXNXXX,1,Dial(SIP/your_external_trunk/${EXTEN})
  same = n,Hangup()

; This is a call that should be transfered back to the secretary's 
extension when external call is finished
exten = _#NXXNXXX,1,NoOp(Special Dial for Boss/Secretary Transfer)
  same = n,Dial(SIP/your_external_trunk/${EXTEN:1},,g)
; First call has ended, now we go back to the secretary)
  same = n,Dial(SIP/1234)
  same = n,Hangup()

That's at least where I would start with my testing and then develop the 
solution from there.-- 
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Re: [asterisk-users] XMPP + Asterisk integration - a practical and simple example

[Kevin Larsen]

 
http://www.mundoopensource.com.br/en_page_xmpp_asterisk_pratical_example/

Wish I had seen this when I was setting it up on my systems. Played around 
quite awhile using something other than OpenFire and couldn't get it 
working no matter what I did. Switched to OpenFire and while it wasn't 
completely smooth sailing, it worked much better.-- 
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Re: [asterisk-users] OT: Question on Caller ID (Spoofing calls with Asterisk)

[Kevin Larsen]

 I got a call from an overseas call center telling me about the
 problems with the Windows machine I was using. They wanted to remote
 in and fix things for me ... (Ignore the fact I use a MacBook Pro or
 an ASUS laptop with Debian).
 
 What I found curious was the caller's name was Asterisk, and the
 caller's number was Asterisk@10 or or Astrk@10 similar. (I don't
 recall the exact number, but it was malformed and it had an '@' in
 it).
 
 I'd like to read a little more about spoofing calls with Asterisk. Can
 anyone provide a reference?

There really isn't much extra to read. Like the others have said, I can 
set my caller id to be anything I want with Asterisk. Whether the 
downstream carrier will accept it is another matter entirely. I work with 
multiple carriers at my locations around the world and have found they 
usually do one of three things. 

1. Allow only the main number on the account as the outbound caller ID. I 
hate this one as I may very well want my CID to not be the main number in 
some cases.
2. Allow the CID to be any number I own through that carrier. This one is 
preferable as it allows people to have their direct dial number show up as 
their caller ID.
3. Allow the CID to be any number. This one is how you get spoofing to 
work. The carriers themselves can still tell who actually sent the call, 
but most people won't go through the hassle of tracking it down to get the 
spoofers taken care of.

Additionally, some carriers will reject an outbound call from you if your 
CID isn't set correctly, others will just silently reset it to your main 
number in the background.-- 
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Re: [asterisk-users] FYI: Block Comments

[Kevin Larsen]

 The configuration parser can do a lot of things. Out of curiosity 
 amongst those reading this - how many of you know about templates?
 

I use templates and wish the realtime parser would understand them as 
well.-- 
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Re: [asterisk-users] Asterisk 12 - queue variables not passed to local channel

[Kevin Larsen]

 Asterisk 12.5
 
 I'm using AMI to initiate a call me now feature from the web site. 
 The AMI looks like:
 Action: Originate
 Channel: Local/s@callmenow
 Context: dial-to-customer
 Exten: s
 Priority: 1
 Async: true
 Variable: CHANNEL_TO_CUSTOMER=SIP/voipms/111222
 Timeout: 99
 
 Dial Plan:
 [callmenow]
 exten = s,1,NoOp(callmenow: Queue without answer)
same =n,Queue(sales,Rtc)
 
 [dial-to-customer]
 exten = s,1,NoOp(dial-to-customer channel=${CHANNEL(name)})
same =n,DumpChan()
 
 The dial-to-customer context is invoked when the sales queue agent 
 answers the phone.
 
 When the local channel is used, the queue related variables, 
 specifically MEMBERINTERFACE, are missing.  When a normal call 
 (typically SIP or DAHDI channel) enters the queue, the MEMBERINTERFACE 
 and other variables are present.
 
 my queues.conf has
 setinterfacevar = yes
 setqueueentryvar = yes
 setqueuevar = yes ;
 
 I didn't see anything in the V12 doc that related to this.
 
 Is this a bug or a feature?

I haven't done what you are looking to do exactly, but I think I 
understand where you are going with this.

Take a look at this link: 
http://www.voip-info.org/wiki/view/Asterisk+local+channels

I think if you add a /n to your local channel, it might do what you want. 
From reading this, it looks like the local channel is being optimized out 
and causing you to lose some of your variables that you had set. So, in 
your AMI, 
change this: Channel: Local/s@callmenow
to this: Channel: Local/s@callmenow/n
and see if that gets you what you are looking for. I bet the local channel 
has the MEMBERINTERFACE variable and it gets lost when optimized out.-- 
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Re: [asterisk-users] Better info on call failure

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 08/13/2014 08:31:01 AM:

 From: Nick Olsen n...@flhsi.com
 To: asterisk-users@lists.digium.com, 
 Date: 08/13/2014 08:31 AM
 Subject: [asterisk-users] Better info on call failure
 Sent by: asterisk-users-boun...@lists.digium.com
 
 Hey everyone,
 
 Currently, I've got a PBX that is emailing me on call failures to an
 international SIP provider of ours.
 
 I'm doing this with exten = 1,1,System(mail -s Call from $
 {CALLERID(num)} to ${DNID} Failed with DialStatus ${DIALSTATUS} 
 n...@flhsi.com  /dev/null)
 
 This works fine, However it's a little lacking. For Instance,
 
 Our INTL SIP provider will bounce back SIP status messages if the 
 call is rejected. 503 Service unavailable. 6XX over rate limit. 6XX 
 blocked destination..etc. Anyone have any ideas about how I might 
 capture that and include it in my email. Right now they just all 
 bounce CHANUNAVAIL which is expected.
 
 Thanks!

You could write a shell script that handles the actual mailing and pass in 
the information as arguments when calling the shell script.

exten = 1,1,System(/opt/scripts/asterisk/send-error-email.sh 
${CALLERID(num)} ${DNID} ${DIALSTATUS} ${HANGUPCAUSE})

In this fashion you could also have a body in your message that contains 
all the extra information. This would work for basically any variable you 
wanted to pull in from Asterisk.-- 
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Re: [asterisk-users] Sending and receiving fax with Digium FFA

[Kevin Larsen]

 Hello. 
 I've been trying to setup Free Fax for Asterisk on a Debian machine 
 with Asterisk 1.8. I have managed to register and installed the 
 Digium modules. Sending and receiving through it have resulted in 
 failure. The output of fax show capabilities is:
 Registered FAX Technology Modules:
 
 Type: DIGIUM
 Description : Digium FAX Driver
 Capabilities: SEND RECEIVE T.38 G.711 MULTI-DOC
 
 1 registered modules
 
 We have a fax blackbox  through which I'm trying to send faxes to 
 the Asterisk server. Every time that I send a fax I get a timeout 
 error. Been tinkering with the settings and whatnot to get it working.
 
 
 
 The extension to receive fax:
 exten = recvfax,1,Verbose(2,Receiving fax)
 same = n,Set(FAXDEST=/tmp/fax)
 same = n,Set(tempfax=${STRFTIME(,,%C%y%m%d%H%M)})
 same = n,Wait(8)
 same = n,ReceiveFax(${FAXDEST}/${tempfax}.tiff,f,d)
 It's without most of the tinkering I've done, which are: setting ecm
 to no, tweaking the min/max rate and other things.
 
 Also, because the fax machine can't print (half broken), we receive 
 our faxes through a fax to email service we have subscribed to, so 
 the tests for sending have that one as a destination. 
 
 The extension to send fax:
 exten = sendfax,1,Verbose(2,Sending fax)
 same = n,Set(faxlocation=/tmp)
 same = n,Set(faxfile=fax.tiff)
 same = n,Set(FAXOPT(headerinfo)=Testing FAX)
 same = n,Set(FAXOPT(localstationid)=123456)
 same = n,SendFax(${faxlocation}/${faxfile},d)
 same = n,Verbose(2, Fax Status: ${FAXOPT(error)})
 I did the exact same thing, and tried sending from both a SIP 
 channel and a DAHDI line. The weird thing is that when I am sending 
 through Asterisk I get, as a response to fax, a recorded message 
 from the telco. Sending through the same line with the fax machine 
 works perfectly.
 
 Any advice and help is welcome.

Can you post the output of the Asterisk CLI from a failed fax call? What 
you have looks ok for the most part, at least on the receiving end. Did 
you install the license key for the Free FAX for Asterisk module?-- 
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Re: [asterisk-users] The plain old PBX functionality

[Kevin Larsen]

 I am not sure why a previous response refers to this module as 
 'toxic'. It is a free to use module which allows a host of Digium 
 phone features to be quickly implemented with Asterisk, like 
 security-enhanced auto provisioning.

Without creating a large off-topic response, there is a segment of the 
open source movement that holds that any software that does not come with 
source code is bad and should not be touched/used in any fashion by any 
person/company.-- 
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Re: [asterisk-users] The plain old PBX functionality

[Kevin Larsen]

 back in the old analog telephony days there was digital PBX-es and
 digital system phonesets. This phonesets have had many individual
 illuminatable buttons connected with extensions. The PBX can show on
 the buttons if some extension is ringing (blinks) or busy (constant
 light), and the user can transfer the call with one touch (pressing
 one of this button).
 
 I search this functionality in Asterisk. What versions, and what
 extension functions (or other settings), and what VoIP phones can do
 this?

Asterisk has had this functionality for a long time. The terms you want to 
search for are BLF (Busy Lamp Field) and Subscribe. I imagine that most 
sip phones have the necessary features to do BLF. I know the Polycom 
phones I use certainly do. The Digium branded phones do as well.-- 
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Re: [asterisk-users] Loud Ringers and paging systems...

[Kevin Larsen]

 if you use a papt2 or so spa2101 then you could have alert info set 
 to different lengths or styles of ringers
 
 i use that in a dorm with phones and have the phones ring short 
 rings at night so it wont wake up the students 

I do not use either of those devices, but after posting this yesterday, I 
did end up coming up with a way to do this making use of Asterisk dialplan 
code and the paging side of the paging/night ringer system.

The basic concept is that the original call will run a script that creates 
a call file to call the paging system and play a specific audio file. It 
also passes into the paging call its channel name. In the call to the 
paging system, I use the SHARED function to write back to the original 
calls' channel the channel name of the paging call. Then when the original 
call is answered, it runs a subroutine that redirects the paging call to a 
priority that hangs that call up.

If anyone is interested, I have my proof of concept code that I could post 
up to the group.-- 
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Re: [asterisk-users] Loud Ringers and paging systems...

[Kevin Larsen]

 
 Will your approach handle ringing more than one of the three 
 extensions simultaneously?
 
   --Don

Not if they are in the same paging zone, but neither would using the night 
ringer function on the pa system, so I consider that acceptable. Not even 
sure what would be considered correct in the case of two at the same time. 
First come/first serve is the only thing that comes to mind as being 
reasonable.-- 
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Re: [asterisk-users] Unregistered ports on SPAxxxx

[Kevin Larsen]

 I've got a few devices, SPA112's and SPA8000's, that are giving me 
problems.
 
 Each device has a separate SIP credential for each port, but 
 sometimes, only a 
 few of the ports register. 
 
 So, the device will be running fine for a while, then suddenly one or 
more of 
 the ports will become Unreachable.  These ports will stay unreachable 
until 
 the device is power cycled.
 
 I'm presuming that there was a momentary interruption in connectivity 
that 
 caused the registrations to fail/timeout.  But the ports should have 
become 
 Reachable by the time the registration period elapses.  But they don't.
 
 Any ideas?

Interesting you should mention this. I have an SPA-112 that is giving me 
fits right now. Multiple times per week it goes down and has to be power 
cycled. When it is down, it is not registered with Asterisk, I cannot 
reach its configuration web page, but I can ping it. Mine is running 1.2.1 
(004) on the firmware, but I see that 1.3.3 (015) is out. That was going 
to be my next change to see if it helps. -- 
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[asterisk-users] Loud Ringers and paging systems...

[Kevin Larsen]

Working on a paging system for one of my sites and running into something 
I can't believe is this hard. In one of the zones, they want to have three 
different extensions ring over the pa system, using it as a loud ringer. 
Now the paging system does have a loud ringer built in and I can easily 
have it do a simultaneous ring, but all of the extensions will sound the 
same over the loud ringer. Of course, we want them to have different rings 
over the pa system so that all three people don't have to check their 
phone every time it rings.

So far, the only semi solution I am coming up with (short of buying three 
different loud ringers and wiring them into the paging system) is to have 
my dialplan generate a call file that will make a second call to the 
paging system and play out an audio file based on who we are doing the 
loud ringer for. This has the disadvantage that it isn't a true loud 
ringer as it will only play for however long I tell it to and it won't cut 
off if they answer the phone before the audio file finishes playing. 

Anyone have any suggestions about a better way to handle this? Really 
hoping there is an Asterisk dialplan solution as I don't want to triple my 
paging hardware just to add one tiny piece of functionality.

Kevin Larsen-- 
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Re: [asterisk-users] Asterisk 12 and DPMA

[Kevin Larsen]

 I read somewhere that DPMA is not supported for Asterisk 12.  Can anyone 

 confirm or deny that?  If not supported yet, will it be? If so, when?

Per this link:
https://wiki.asterisk.org/wiki/display/DIGIUM/Digium+Phone+Module+for+Asterisk+(DPMA)+v+2.0

It would seems that Digium is under the impression that it is supported.-- 
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Re: [asterisk-users] Simultaneous Ring

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 07/16/2014 01:46:09 PM:

 From: Haley,Scott A scott.ha...@edwardjones.com
 To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com, 

 Date: 07/16/2014 01:46 PM
 Subject: [asterisk-users] Simultaneous Ring
 Sent by: asterisk-users-boun...@lists.digium.com
 
 I have a need to issue a dial command to a number:
 
 same = n,Dial(${DIALGROUP1},${TIMER1},t)
 
 After a number of seconds, let's say 10 seconds. I want to dial 
 another set of numbers while continuing to ring, or interrupting the
 first group of numbers.
 
 same = n,Dial(${DIALGROUP2},${TIMER1},t)
 
 Is there a way to do this without interrupting the first call?
 
 Thanks,
 Scott Haley

I believe that what you want to do is best done with Local Channels. See 
this link:
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeeperDialplan_id324598.html
for more information.-- 
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Re: [asterisk-users] How to monitor non-SNMP SIP devices ?

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 07/09/2014 10:19:11 AM:

 From: Olivier oza.4...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com, 
 Date: 07/09/2014 10:19 AM
 Subject: [asterisk-users] How to monitor non-SNMP SIP devices ?
 Sent by: asterisk-users-boun...@lists.digium.com
 
 Hi,

 I'm seeing a trend in which SIP devices such as Yealink SIP phones 
 (with v72 firmware), are dropping support of SNMP in favor of HTTP 
 eventing if may call this as such :
 when configuring the SIP device, you can define a couple of HTTP URL
 which triggered when some event occur (end of boot, on hook, ...).

 How do deal with those devices ?
 Do you still try to monitor them with usual tools (Nagios, OpenNMS) 
 or do you favor another  class of software ?

We don't monitor our phone endpoints (we do our trunks), but if I were to, 
I would probably set up a simple webserver with some php that would write 
the logs to a sql database. 

What you describe isn't really as good as snmp though, because I can have 
my monitoring system poll snmp devices, whereas HTTP eventing depends on 
an event happening to trigger the contact. If the phone goes down hard or 
locks up, I may not know there is a problem or just no events have 
happened. I hope at the least, they have a keep alive event that can 
periodically access the url to indicate all is well. 

On things I want to monitor, I just don't like the idea of not being able 
to have my monitoring system talk to them and depending on them talking to 
my monitoring system. That would probably make me heavily reconsider 
buying any more of their products if it was something I depended on.-- 
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Re: [asterisk-users] share mailbox Asterisk 1.8.22

[Kevin Larsen]

I have done this for one of my users in a very similar fashion. When 102 
checks the voicemail, do they hear the correct voicemails? Ours clears 
just fine in this situation.

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208

asterisk-users-boun...@lists.digium.com wrote on 06/24/2014 04:37:26 PM:

 From: motty cruz motty.c...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com, 
 Date: 06/24/2014 04:37 PM
 Subject: [asterisk-users] share mailbox Asterisk 1.8.22
 Sent by: asterisk-users-boun...@lists.digium.com
 
 Hello, I want to share mailbox between two extensions
 Ext. 101
 Ext. 102
 
 I want the messages to go to mailbox 101, when when checked mailbox 
 from extension 102 to be able to clear the bliking red light. 
 
 here is extensions.conf
 exten = 102,hint,SIP/${EXTEN}
 exten = 102,1,Dial(SIP/101SIP/102,20,t)
 exten = 102,2,Voicemail(101,u)
 exten = 102,102,Voicemail(101,b)
 exten = 102,103,Hangup
 
 sip.conf
 [102]
 type=friend
 context=sipphones
 call-limit=99
 callerid=Jo 102
 disallow=all
 allow=ulaw
 allow=alaw
 username=102
 secret=xexpasswd
 dtmfmode=rfc2833
 host=dynamic
 mailbox=101
 nat=yes
 canreinvite=no
 
 this configuration work ok but, light msg keeps bilking after 
 checking for mesages. 
 
 Any suggestions?
 Thanks, 
 
 
 __
 This email has been scanned by the Symantec Email Security.cloud 
service.
 For more information please visit http://www.symanteccloud.com
 __-- 

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Re: [asterisk-users] share mailbox Asterisk 1.8.22

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 06/24/2014 05:36:16 PM:

 From: motty cruz motty.c...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com, 
 Date: 06/24/2014 05:36 PM
 Subject: Re: [asterisk-users] share mailbox Asterisk 1.8.22
 Sent by: asterisk-users-boun...@lists.digium.com
 
 yes they're able to hear the same msg, in /var/spool/asterisk/default/
 rm -rf 102
 ln -s 101 102
 
 but it does not clear out, 
 
 Thanks, 


That is where your setup and mine are different. I have my second 
extension directly check the first extensions mailbox as opposed to using 
a symlink. That way, when the box is cleared, it is actually happening in 
the original mailbox.

Basically, my code to check voicemail uses the CALLERID(num) to determine 
the mailbox and I have both extensions set to use the same caller ID. This 
works for me as both extensions belong to the same person (one phone at 
the office, one at home) and they want to always have their outbound calls 
show up as if they were from their published number regardless of which 
phone they use.-- 
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Re: [asterisk-users] share mailbox Asterisk 1.8.22

[Kevin Larsen]

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208

asterisk-users-boun...@lists.digium.com wrote on 06/24/2014 05:49:39 PM:

 From: motty cruz motty.c...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com, 
 Date: 06/24/2014 05:49 PM
 Subject: Re: [asterisk-users] share mailbox Asterisk 1.8.22
 Sent by: asterisk-users-boun...@lists.digium.com
 
 Thanks Kevin, 
 
 can you provide me with example of your code? if you don't mind? 
 
 Thanks,
 

Sure, Here is the relevant code (I have removed parts that don't apply):

sip.conf:

[501]
callerid=Bob Smith 501
secret=501
mailbox=501

[502]
callerid=Bob Smith 501
secret=502
mailbox=501


voicemail.conf:

[default]
501 = 1234,Bob Smith


extensions.conf

exten = *99,1,VoiceMailMain(${CALLERID(num))
  same = n,Hangup()

exten = 501,1,Dial(SIP/501SIP502,16)
  same = n,VoiceMail(${EXTEN},${IF($[${DIALSTATUS} = BUSY]?bd:ud)})
  same = n,Hangup()


This isn't a complete dialplan, but it is the simplest way I could break 
mine down to just the relevant parts. When you hit the voicemail button on 
my phones, they dial *99 to access voicemail. By setting extensions 501 
and 502 to both use 501 as the caller id number, either phone logs into 
mailbox 501. There are other ways you could approach this in the dialplan, 
but this is the one that worked out for me.-- 
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Re: [asterisk-users] second connected PBX not showing Caller ID

[Kevin Larsen]

 From: Claude Hayn chayn...@gmail.com
 To: asterisk-users@lists.digium.com, 
 Date: 05/31/2014 04:43 PM
 Subject: [asterisk-users] second connected PBX not showing Caller ID
 Sent by: asterisk-users-boun...@lists.digium.com
 
 Hello,
 
 We have two asterisk PBXs connected. 
 PBX 1 has SIP trunks connected to our provider.  PBX 2 is a remote 
 PBX and SIP Trunk connected to PBX 1. 
 We are able to dial extensions either way and PBX 2 is able to dial 
 out using PBX 1  SIP trunks connected to our provider.
 
 We would like to use a separated Caller-ID for PBX 2 and cannot 
 figure out how to do this.
 
 Any suggestions would be greatly appreciated.
 
 Thank you,
 
 Claude 

Claude,

Without seeing your dialing plan, it is hard to say for certain what you 
should do. However, there are a couple of things you could check.

1. When a call comes in from PBX 2 to PBX 1, does it have the caller ID 
you want already set? If so, then something in PBX 1 is overwriting it. 
The way to handle this is to have a separate path set up for external 
calls that come in from PBX 2. That way you can ensure that your caller ID 
isn't getting clobbered. If you are sure that you are setting the CID 
correctly before the call goes out, look at the next item.

2. Does your SIP provider allow you to set your caller ID? I have seen 
three answers to this question. Some allow you to set the CID to any 
number you want, even if it isn't a valid number or one you own. Some 
allow you to set it to any number, as long as it is one you own through 
them. Finally, some of them let you pick one number and all calls get that 
CID regardless of what you set before you send the call to them. If your 
provider falls into the third category, you will need to contact them to 
find out if you can be allowed to set your CID to a different number. I 
had a provider for one of my locations that defaulted to one number only, 
but was happy to make a change to allow me to set it to whatever number I 
wanted out of the ones they provided service for.

If neither of these work, we would likely need more details to be able to 
help spot the problem.-- 
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Re: [asterisk-users] 'restart when convenient'

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 05/28/2014 10:37:25 AM:

 pbx1*CLI core restart when convenient 
 Waiting for inactivity to perform restart
 Ignoring asterisk restart request, already in progress.
 
 
 After doing 'core restart now' and hitting Enter really hard ;) Asterisk
 did restart.
 
 
 Some how Asterisk thinks it is not convenient. I want to find out why.
 

I haven't had it fail to restart, but I have been in the same situation 
and had it have a nice delay of a minute or two before it finally finds it 
convenient to restart. Haven't figured out what the delay is though. I am 
on 11.6.0.-- 
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Re: [asterisk-users] BLF and notifyringing in Asterisk 11

[Kevin Larsen]

 Unfortunately, notifyringing is only set in the [general] section in
 sip.conf. It does not have a peer level override.
 
 It would be nice if it was set on a peer by peer basis - that would be
 a useful improvement.
 
 -- 
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org
 

Matt,

Can you clarify exactly what it should do in the case of a queue call? 
Again, my scenario is that I have a queue that is using the ringall 
strategy. I have ringinuse enabled as the main person may want to end 
their call and pick up the new call. However, when I set notifyringing to 
no, if no one is on a call, the phones indicate ringing as normal. If even 
one of the people in the queue are on a call, all of the phones light up 
as in use instead of the ones on a call being in use and the others 
showing ringing.

Based on what you said above, I suspect that the code is looking at all 
the phones that are being run for this call and deciding on a global basis 
how to indicate the ringing/in use status. Would that be correct?

Short of a code change, is there any way you can see to do what I want?-- 
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[asterisk-users] BLF and notifyringing in Asterisk 11

[Kevin Larsen]

I am trying to get something working that is just not doing quite what I 
want. It may not be possible, but I figured it was worth asking about. 

The details:
Asterisk 11.6.0
Polycom SoundPoint IP650 phones running 4.03 firmware.

We have a queue with 4 phones in it. ringinuse is set to yes and the 
stategy is ringall. In sip.conf, we have notifyringing set to yes as well. 
Asterisk is sending messages of the type application/dialog-info+xml to 
the phones.

This works nicely in almost every scenario. We have one person on the 
queue who answers the phones first, the rest of us only pick up if he is 
already on another call and not picking up. We have ringinuse set to yes 
because there are many times that he will be able to end his current call 
to pick up the new one, so we want to keep this setting. However, if a 
call comes into the queue and he is on another line, we would like his blf 
light to stay at the inuse value (red on the polycom) and not the ringing 
value (flashing green on the polycoms).

Now the problem. If I set notifyringing=no on the sip definition for his 
extension, it doesn't seem to get applied. If I set notifyringing=no in 
the general section, then it does get applied. However, if I put it in the 
general section, then none of the phones in my queue ever show a ringing 
state. When they are ringing, they show the solid red light of the in use 
state.

What I would like is to see the following happen:
If no one is on a call, all phones show ringing on their respective BLFs.
If one phone is on a call and a second comes in, the phone on the call 
stays in use (solid red) and the rest show ringing (flashing green). So 
far, no matter what combinations of notifyringing I use, I can only get 
either all the phones to show ringing or all of them to show in use.

The state being sent to the polycom is 'early' for a ringing phone and 
'confirmed' for an in use phone. In the case of a phone that is both in 
use and ringing, I get a state of 'confirmed' followed immediately by a 
state of 'early'. This is all with notifyringing set to yes. If I read the 
description of notify ringing correctly in the sample sip.conf file, it 
seems like setting it to no should work, but it does not. 

;notifyringing = no ; Control whether subscriptions already 
INUSE get sent
; RINGING when another call is sent 
(default: yes)

Not sure if this rises to the level of a bug or is just my 
misunderstanding of how this should work. With the description above, I 
would expect that setting notifyringing to no would mean that I get the 
early state if the phone isn't already on a call, but would remain at a 
confirmed state if a second call came in while already on a call.

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208-- 
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Re: [asterisk-users] authoritative sql definitions for Asterisk Realtime Architecture ARA

[Kevin Larsen]

 Here are links to the Asterisk Wiki for CDR and SIP tables.  I 
 didn't find extensions listed, but it's pretty simple and I can 
 provide the structure for that if needed, but it would be without a 
 definitive source beyond me having used it for years.  :-)

I think the problem with those links are that they are as close as you get 
to authoritative, but they are not complete nor totally correct.

Two examples I can think of off the top of my head are that the sendrpid 
enum definition only has yes or no. pai should also be a valid option. 
Second, there is no description column. It should be a varchar(40).

Probably the only way to make a definitive list for this would be to find 
the appropriate source files and look at every place they can read in from 
the database to see what columns they can read.-- 
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Re: [asterisk-users] Multicast RTP

[Kevin Larsen]

 From: Josh Metzger joshdmetz...@gmail.com

 I'm currently working with Asterisk 11.8.1 trying to get Multicast 
 RTP working (it's not) with some Polycom phones, and I'm really 
 trying to determine if Asterisk or the phones are the issue.  I 
 THINK it's Asterisk...

 In extensions.conf I have a simple: Page(MulticastRTP/basic/
 x.x.x.x:) line, and when I dial that extension I get:
 
-- Called MulticastRTP/basic/x.x.x.x:
 

   
 -- MulticastRTP/0x7f8b4000f898 answered SIP/XXX-004c 

 After connecting and hearing the beep the line stays open and I 
 can talk and press buttons and so on, but the phones aren't getting 
 anything.  I ran rtp set debug on and if I call extension to 
 extension I see all of the got RTP packet from and Sent RTP 
 packet to messages as expected, but doing the same thing when 
 calling my Multicast Page extension only shows me Got RTP packet 
 from messages.  Shouldn't I see the Sent RTP packet to messages 
 with the Multicast address/port displayed?  I've also run a 
 wireshark capture and all I see is the RTP stream from my phone to 
 the server - nothing going back out.  What am I missing, here?

See here: 
http://community.polycom.com/t5/VoIP/Asterisk-1-8-Multicast/td-p/10918

It refers to Asterisk 1.8, but the situation remains the same. Polycom 
phones, to my knowledge, do not work with any kind of multicast stream 
that is supported by Asterisk. They need the whole SIP signalling to set 
up the call. We use Polycom phones and the way we worked it out was to 
build a dialgroup with all the active phones and then page that dialgroup.

Here is the code I am using:

exten = s,1,SIPAddHeader(Alert-Info: Ring Answer)
  same = n,Gosub(sub_active_phones,${EXTEN},1(page))
  same = n,Set(CALLERID(name)=Emergency Page)
  same = n,Page(${DIALGROUP(page)},is)
  same = n,Hangup()

The sub-routine I call goes through all our extensions and builds a 
dialgroup of only those that are currently reachable and not on a call.
On the Polycom side, they are set to auto answer when they see the 
Alert-Info: Ring Answer header. Yes, this does mean that I am generating 
one call for every phone I am paging and yes it is less ideal (by far) 
than using multicast rtp. We did tests to determine that in an emergency 
it put an acceptable load on Asterisk and that it wouldn't cause it all to 
crash and burn. -- 
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Re: [asterisk-users] Multicast RTP

[Kevin Larsen]

 From: Josh Metzger joshdmetz...@gmail.com

 Interesting.  I thought the latest Polycom software supported 
 multicast, but that Polycom forum link says otherwise.  What DOES 
 work is using the built-in paging feature, so maybe the solution, in
 this case, is to do it without Asterisk at all.  We currently have a
 setup similar to what you have which works, but isn't as optimal as 
 doing it multicast (lots of phones leads to the message getting 
 chopped for some phones).  In any case, thanks for the info!

If I recall correctly, the only reason we didn't like the built in paging 
feature is that it would put a paging soft button on every phone where we 
enabled it. It was unacceptable to the powers that be to have that button 
there, but we still needed to be able to page from all the phones in an 
emergency. Thus we went with the Asterisk paging solution using a 
dialgroup. In our setup we are paging around 100 phones and everything is 
able to stand up to the load. A much larger setup, though, and it likely 
would not work as well. It does take our Asterisk server to between 40 and 
60 per cent cpu usage while the paging is occurring, where it normally 
runs less than 5%. Audio quality remains normal. As it is emergency only, 
that was deemed acceptable.-- 
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Re: [asterisk-users] Putting a notice in the logs from the dialplan

[Kevin Larsen]

 From: Matthew Jordan mjor...@digium.com

 Ha! Just when you think you've found every corner of Asterisk, you
 turn around and there's something else.
 
 Just goes to show, you learn something new every day.

Look on the bright side, you did say it would be easy to write just such a 
module...-- 
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Re: [asterisk-users] FW: clients unable to auth

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 04/16/2014 05:56:32 AM:

 From: Peter Reid peter.r...@morodo.co.uk
 To: asterisk-users@lists.digium.com, 
 Date: 04/16/2014 05:56 AM
 Subject: [asterisk-users] FW: clients unable to auth
 Sent by: asterisk-users-boun...@lists.digium.com
 
 Hi Guys,
 
 Just new to Asterisk and am completely stumped.  I have created two 
 accounts as instructed.  Please see below for the config of the user
 accounts. 
 
 [Peter]
 type=friend
 host=IP address
 disallow=all
 allow=ulaw
 allow=alaw
 callerid=Peter 6004
 secret=XXX
 context=default
 port=9060
 nat=force_rport,comedia
 deny=0.0.0.0
 permit=IP/255.255.0.0,IP/255.255.0.0,PrvIP/255.255.0.0, 
PrvIP/255.255.0.0
 

Your phone is registering with the name 6004 and not Peter. You either 
need to change [Peter] to [6004] in Asterisk or update your phone config 
to make it use Peter for your authorization name.-- 
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Re: [asterisk-users] DUNDi with SIP Mapping

[Kevin Larsen]

 From the reading and testing I have done it doesn't look like SIP 
 supports a username and password in the Dial string. I currently 
 have the following mapping.
 
 priv = dundi-extens,0,SIP,dundi:pass@1.1.1.1/$
 {NUMBER},nounsolicited,nocomunsolicit,nopartial

 On the sending side I see
 
 NOTICE[31598] chan_sip.c: Conflicting extension values given. Using 
 'dundi' and not '1001'

 On the receiving side it will not match the SIP dundi user and tries
 to call dundi instead of 1001.
 
 -- Executing [dundi@from-sip-external:1] NoOp(SIP/1.1.1.
 2-, Received incoming SIP connection from unknown peer to 
 dundi) in new stack
 

 Is there a way to configure DUNDi to use SIP or does it only work with 
IAX?

I am using DUNDi with SIP to do some least cost routing amongst my various 
locations. My mapping is close to what you have:

priv = dundi-extens,0,SIP,trunk_name/number_to_dial

Where trunk_name is replaced with the actual name of my trunk as defined 
in sip.conf and number_to_dial is the number they should dial on that 
trunk. I have not tried to define the SIP username/password in the DUNDi 
config itself, so I don't know if what you are trying to do is possible or 
not.-- 
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Re: [asterisk-users] FW: clients unable to auth

[Kevin Larsen]

 Thank you guys – your advice was spot on.  I will now reach out 
 earlier and not struggle with issues like this for 2 weeks J 

You sound like you are just getting started with Asterisk. A couple pieces 
of advice that helped me when I was starting out: 

1. Get a copy of Asterisk: The Definitive Guide. Work through the examples 
and understand the concepts it teaches. I still use it all the time.
2. When you run into problems, http://voip-info.org is a great Asterisk 
resource. It isn't always perfectly up to date, but is very useful.
3. Search on the error messages you are given by Asterisk. It is common 
enough that many (but not all) of the error messages will have hits that 
explain your problem in greater detail.
-- 
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Re: [asterisk-users] DUNDi with SIP Mapping

[Kevin Larsen]

 I wanted to move to DUNDi to simplify the setup. It looks like I 
 need to switch to IAX trunks to be able to do this.

You are a bit outside of what I have done, but this looks like it might be 
what you want to do with SIP:
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP-- 
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Re: [asterisk-users] Need to hire recordings for an IVR

[Kevin Larsen]

 I wonder if anybody know how to hire Alice or some professional
 voice-artist. I need to record 12 messages for a customer.

Assuming you mean Allison, her information is here:
http://www.digium.com/en/products/ivr/allison-smith-- 
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Re: [asterisk-users] Asterisk 11 under VMware?

[Kevin Larsen]

 From: Johan Wilfer li...@jttech.se
 Sounds very good. Do you have this experience with WMware in particular 
 or with virtualization in general?

We run our Asterisk 11 instance in VMWare as well. They share the hardware 
with multiple other boxes. We do give Asterisk priority over most other 
virtual machines. We either have SIP providers or use boxes like Digium's 
G100 series to convert our T1 lines to SIP.

Our experience has been good and we have no problems loading Asterisk up 
on virtual machines on each site.-- 
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Re: [asterisk-users] Asterisk CLI Banner

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 03/28/2014 10:51:13 AM:

 From: Haider Khalil haiderkha...@hotmail.com
 
 Thank you Thorsten Göllner.
 
 Matthew, 
 
 What does violating license of Asterisk means ? Does it means I 
 won't be able to use any commercial modules or asterisk commercially
 ? I thought it was open and anyone can change the code ?
 
 Haider

I am neither a lawyer or a licensing expert, but the basics are that if 
you make such a change for your own internal use, you are probably fine. 
Example: You have 10 sites with Asterisk in them and at each site you have 
someone in your company who has to log into the CLI and do stuff. You 
change the header to pass them a message. This is probably (not going to 
guarantee this) going to be fine as it is not something you are releasing 
out into the wild nor are you selling it and making a profit from it.

However, let's say you make a commercial project that uses Asterisk under 
the hood and you change the header to hide the fact that it uses Asterisk 
and that Digium has any ownership of the code. That would be not be okay 
in most, if not all cases.

Basically, the code is open source, but it is still owned by Digium and 
they have specific rights that you have to be careful of in regards to 
licensing. If someone outside of your organization will ever be running 
the code you change, there are specific rules that have to be followed, 
including those that relate to releasing your changes to the code and to 
giving credit back to those who wrote the code your code is based on.

Basically, Richard Kenner is spot on. If you are unclear, best to consult 
an attorney who specializes in this, especially if you are redistributing 
the altered code.-- 
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[asterisk-users] XMPP issues in Asterisk 11.6.0 for distributed device states...

[Kevin Larsen]

I have been working with distributed device states in Asterisk using XMPP 
attached to an OpenFire server. I have it working well across two servers 
and want to roll it out across every server in my company. All servers are 
Asterisk 11.6.0. I am running into a problem that seems like it should be 
a bit easier to solve than it is seeming to be. On the third server I am 
rolling into this solution, I get plenty of the following:

res_xmpp.c:1398 xmpp_pubsub_handle_error: Error performing 
operation on PubSub node device_state, 403.

So, basically, servers 1 and 2 continue to hum along nicely updating their 
device state, but server 3 gets a 403 forbidden message when it tries to 
deal with device state. I believe this has to do with the permissions set 
up on the device state node. I have a small example that demonstrates the 
creation of a new node.

In the Asterisk CLI, I ran 'xmpp create collection asterisk test' on 
server 3, which was successful and can be seen on servers 1 and 2 with 
'xmpp list nodes asterisk'

The debug output from server 3 for this is as follows:

--- XMPP sent to 'asterisk' ---
iq to='pubsub.xmpp' from='server3@xmpp/astvoip3' type='set' id='aaacy'
  pubsub xmlns='http://jabber.org/protocol/pubsub'
create node='test'/
configure
  x xmlns='jabber:x:data' type='submit'
field var='FORM_TYPE' type='hidden'
  valuehttp://jabber.org/protocol/pubsub#owner/value
/field
field var='pubsub#node_type'
  valuecollection/value
/field
field var='FORM_TYPE' type='hidden'
  valuehttp://jabber.org/protocol/pubsub#node_config/value
/field
field var='pubsub#deliver_payloads'
  value1/value
/field
field var='pubsub#persist_items'
  value1/value
/field
field var='pubsub#access_model'
  valuewhitelist/value
/field
  /x
/configure
  /pubsub
/iq
-

--- XMPP sent to 'asterisk' ---
iq to='pubsub.xmpp' from='server3@xmpp/astvoip3' type='set' id='aaacz'
  pubsub xmlns='http://jabber.org/protocol/pubsub#owner'
affiliations node='test'
  affiliation jid='server1@xmpp' affiliation='owner'/
  affiliation jid='server2@xmpp' affiliation='owner'/
  affiliation jid='server1@xmpp/astvoip1' affiliation='owner'/
  affiliation jid='server2@xmpp/astvoip2' affiliation='owner'/
/affiliations
  /pubsub
/iq
-

As we can see, the first message creates the test node and sets the access 
model to whitelist, so only jids in the whitelist are allowed to modify 
it. The second message then sets the appropriate server 1 and server 2 
jids to be owners, thus meeting the requirements of the whitelist.

Since these nodes are persistent, it would appear that server 3 cannot 
properly access device_state because it was never whitelisted when the 
node was created originally. I am fairly certain that I can solve this by 
deleting all my nodes and letting them be recreated, but that seems 
extreme as I put more servers into the system. Any thoughts on a better 
way to handle xmpp and making sure new servers can access the proper 
nodes? 


Kevin Larsen - Systems Analyst - Pioneer Balloon Company-- 
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