[asterisk-users] DECT client adapter

[Sebastian Nielsen]

I asked this question previously on list, but never got any reply. So
retrying again.

 

Does anyone know of a DECT client USB module, or DECT client module, or
SIP-to-DECT adapter?

Have searched whole internet, and ONLY found adapters and modules that can
act as a base station. Not a adapter or module that can act as a handset.

 

What I want to do, is that I have a base station, which are locked to 1
handset, for which I cannot access its corresponding SIP details - they are
provisioned from the operator and operator refuses to give SIP credentials.

 

To get around this and still connect a PBX to this, with more handsets and
even desk phones.

 

The idea is then to have some SIP-to-DECT adapter, or USB DECT GAP Phone
adapter, or DECT raspberry pi GAP Phone module.

That will act as a handset. Ergo "register to base station".

 

Then I want my asterisk installation, to use this "Dect handset" as upstream
operator, ergo call outgoing, and receive calls via this "Dect handset"
module.

 

 

Any ideas on how to accomplish this? Anyone that knows of a USB DECT
adapter, than can be switched into "Phone mode", and then be registred into
a DECT base station, that works with Asterisk?

 

 

Best regards, Sebastian Nielsen



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Re: [asterisk-users] STIR/SHAKEN

[Sebastian Nielsen]

Its just that it seems so unrealistic.. WHAT do you need 1M DID’s for? Give 
each stone in your company driveway a own phone number?

1M DID’s = Thats 10% of the population of the country I live in. (sweden)

 

1M DID’s is also three times more than the amount of customers the phone 
operator ”tre” ( https://www.tre.se ) has in sweden, one of sweden’s largest 
phone operators, they are 4th the largest phone operator. (1: Telia, 2: Tele2, 
3: Telenor, 4: Tre)

 

Then you understand why I wonder WTF people are doing… 

 

Best regards, Sebastian Nielsen

 

Från: asterisk-users-boun...@lists.digium.com 
 För d...@donkelly.biz
Skickat: den 12 mars 2021 03:14
Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' 

Ämne: Re: [asterisk-users] STIR/SHAKEN

 

You said it in your first post when you said “I reallt don’t understand.” You 
don’t understand the business that these people are in. A few people showed you 
a few examples of why it’s important to use more than one carrier--and there 
are other reasons that stir/shaken is a big deal for some of us.

 

It clearly isn’t a big deal for you, so you probably don’t have much to add to 
the discussion.

 

--Don

 

 

From: asterisk-users mailto:asterisk-users-boun...@lists.digium.com> > On Behalf Of Sebastian 
Nielsen
Sent: Thursday, March 11, 2021 7:21 PM
To: 'Mailing List' mailto:asterisk-users@lists.digium.com> >
Subject: Re: [asterisk-users] STIR/SHAKEN

 

1:  1M DID’s? Then I would go straight out and say you are a phone operator, 
and then getting your own STIR/SHAKEN certificate shouldn’t be a problem at 
all. Thats a massive amount of numbers, unrealistically many numbers for any 
company ever except for those that are a phone operator.

 

2: For me, its seems like hunting for nano-cents. I checked around when I got 
my DID and call account for my own personal use, and the prices aren’t that 
different. Its really not worth the effort for what you save. Checked with 
several operators and the prices are almost the same per minute, its like one 
operator has like 0.016 per minute and another has 0.014 … not gonna save much 
on that. Might save like 1$-2$ per month on choosing the latter operator.

 

3: Why? Consolidiate all your agreements to 1 single operator that handles 
everything, and everything will be so much simpler. Then you are simply a trunk 
ccustomer to that particular operator, no need to handle all this with signing 
and certificates and everything..

To save a little tiny nano-cent from each minute of call..

 

Från: asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com>  
mailto:asterisk-users-boun...@lists.digium.com> > För Joel Serrano
Skickat: den 12 mars 2021 01:52
Till: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com> >
Ämne: Re: [asterisk-users] STIR/SHAKEN

 

Hi, 

 

I wanted to add some comments to Sebastian's response:

 

1- When you have a lot of DIDs, you can't just "port" them over from company1 
to company2. Try to have 1M or so DIDs and ask if you can just port them. No 
no, not that simple. There is a process that a lot of times is not worth the 
cost/risk/etc.

2- What happens if company1 has very good pricing for DIDs, but extremely high 
rates for placing outbound calls, and company2 has super aggressive pricing for 
the destinations you use most, but sells DIDs very expensive? Mix and match? :)

3- What do you do, when instead of having 1 outbound carrier, you have several 
50? 

 

At the end I think you are mistakenly comparing apples to oranges, your DID 
provider has nothing to do with your outbound carrier, can the DID provider 
also give you outbound calling? Most likely, but that doesn't mean that the 
best way to go is to route outbound calls via the carrier that is providing you 
DIDs.

 

On Thu, Mar 11, 2021 at 4:34 PM Sebastian Nielsen mailto:sebast...@sebbe.eu> > wrote:

I reallt don’t understand why people simply use the same operator to terminate 
your calls, which also provide DIDs for you.

 

Then you don’t need to touch this at all, your carrier will do all the 
STIR/SHAKEN handling for you, you are just a PBX customer.

And then the operator then simply limits your account to only present your DID 
as outgoing number.

 

Seems to be a unneccesary complicated solution just to have your numbers at 
company 1 and have your call termination at company 2.

So fricking unneccessary.

 

What I know there is requirements of number portability, so as long as company 
2 can handle DIDs (ergo ”own” DIDs) you should be able to move your DIDs from 
company 1 to company 2 – then company 2 owns your DIDs.

 

Best regards, Sebastian Nielsen

 

Från: asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com>  
mailto:asterisk-users-boun...@lists.digium.com> > För Alexander Perkins
Skickat: den 12 mars 2021 01:23
Ti

Re: [asterisk-users] STIR/SHAKEN

[Sebastian Nielsen]

1:  1M DID’s? Then I would go straight out and say you are a phone operator, 
and then getting your own STIR/SHAKEN certificate shouldn’t be a problem at 
all. Thats a massive amount of numbers, unrealistically many numbers for any 
company ever except for those that are a phone operator.

 

2: For me, its seems like hunting for nano-cents. I checked around when I got 
my DID and call account for my own personal use, and the prices aren’t that 
different. Its really not worth the effort for what you save. Checked with 
several operators and the prices are almost the same per minute, its like one 
operator has like 0.016 per minute and another has 0.014 … not gonna save much 
on that. Might save like 1$-2$ per month on choosing the latter operator.

 

3: Why? Consolidiate all your agreements to 1 single operator that handles 
everything, and everything will be so much simpler. Then you are simply a trunk 
ccustomer to that particular operator, no need to handle all this with signing 
and certificates and everything..

To save a little tiny nano-cent from each minute of call..

 

Från: asterisk-users-boun...@lists.digium.com 
 För Joel Serrano
Skickat: den 12 mars 2021 01:52
Till: Asterisk Users Mailing List - Non-Commercial Discussion 

Ämne: Re: [asterisk-users] STIR/SHAKEN

 

Hi, 

 

I wanted to add some comments to Sebastian's response:

 

1- When you have a lot of DIDs, you can't just "port" them over from company1 
to company2. Try to have 1M or so DIDs and ask if you can just port them. No 
no, not that simple. There is a process that a lot of times is not worth the 
cost/risk/etc.

2- What happens if company1 has very good pricing for DIDs, but extremely high 
rates for placing outbound calls, and company2 has super aggressive pricing for 
the destinations you use most, but sells DIDs very expensive? Mix and match? :)

3- What do you do, when instead of having 1 outbound carrier, you have several 
50? 

 

At the end I think you are mistakenly comparing apples to oranges, your DID 
provider has nothing to do with your outbound carrier, can the DID provider 
also give you outbound calling? Most likely, but that doesn't mean that the 
best way to go is to route outbound calls via the carrier that is providing you 
DIDs.

 

On Thu, Mar 11, 2021 at 4:34 PM Sebastian Nielsen mailto:sebast...@sebbe.eu> > wrote:

I reallt don’t understand why people simply use the same operator to terminate 
your calls, which also provide DIDs for you.

 

Then you don’t need to touch this at all, your carrier will do all the 
STIR/SHAKEN handling for you, you are just a PBX customer.

And then the operator then simply limits your account to only present your DID 
as outgoing number.

 

Seems to be a unneccesary complicated solution just to have your numbers at 
company 1 and have your call termination at company 2.

So fricking unneccessary.

 

What I know there is requirements of number portability, so as long as company 
2 can handle DIDs (ergo ”own” DIDs) you should be able to move your DIDs from 
company 1 to company 2 – then company 2 owns your DIDs.

 

Best regards, Sebastian Nielsen

 

Från: asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com>  
mailto:asterisk-users-boun...@lists.digium.com> > För Alexander Perkins
Skickat: den 12 mars 2021 01:23
Till: asterisk-users@lists.digium.com <mailto:asterisk-users@lists.digium.com> 
Ämne: Re: [asterisk-users] STIR/SHAKEN

 

Hi Jeff.  What exactly do you mean by the 'inbound piece'?  I've spent quite a 
lot of time with the folks at TILTX understanding the framework; but I am not 
exactly sure what you mean by the 'inbound piece.

 

Greg/Doug, like many folks here, we use LCR.  So, the terminating carrier is 
not necessarily the one that issued us the telephone numbers.  So, they will 
not sign it or simply cannot sign it.  Remember that a very limited number of 
companies can actually sign the calls; the rest have to buy it from these 
'Service Providers'.  

 

And there is another situation - the company you purchase your numbers from and 
the company you place your calls through may be different and both may not be 
able to sign your calls.  Again, a very limited number of service providers 
that can actually sign your calls.  So what do you do in that scenario?  You 
have to find a Service Provider that can:

 

1.  Verify you own that telephone number(s).

2.  Sign your calls.

3.  Provide you with the technical means to do so.

 

So, that's that...  I hope this makes sense.  

 

Alex

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Re: [asterisk-users] STIR/SHAKEN

[Sebastian Nielsen]

I reallt don’t understand why people simply use the same operator to terminate 
your calls, which also provide DIDs for you.

 

Then you don’t need to touch this at all, your carrier will do all the 
STIR/SHAKEN handling for you, you are just a PBX customer.

And then the operator then simply limits your account to only present your DID 
as outgoing number.

 

Seems to be a unneccesary complicated solution just to have your numbers at 
company 1 and have your call termination at company 2.

So fricking unneccessary.

 

What I know there is requirements of number portability, so as long as company 
2 can handle DIDs (ergo ”own” DIDs) you should be able to move your DIDs from 
company 1 to company 2 – then company 2 owns your DIDs.

 

Best regards, Sebastian Nielsen

 

Från: asterisk-users-boun...@lists.digium.com 
 För Alexander Perkins
Skickat: den 12 mars 2021 01:23
Till: asterisk-users@lists.digium.com
Ämne: Re: [asterisk-users] STIR/SHAKEN

 

Hi Jeff.  What exactly do you mean by the 'inbound piece'?  I've spent quite a 
lot of time with the folks at TILTX understanding the framework; but I am not 
exactly sure what you mean by the 'inbound piece.

 

Greg/Doug, like many folks here, we use LCR.  So, the terminating carrier is 
not necessarily the one that issued us the telephone numbers.  So, they will 
not sign it or simply cannot sign it.  Remember that a very limited number of 
companies can actually sign the calls; the rest have to buy it from these 
'Service Providers'.  

 

And there is another situation - the company you purchase your numbers from and 
the company you place your calls through may be different and both may not be 
able to sign your calls.  Again, a very limited number of service providers 
that can actually sign your calls.  So what do you do in that scenario?  You 
have to find a Service Provider that can:

 

1.  Verify you own that telephone number(s).

2.  Sign your calls.

3.  Provide you with the technical means to do so.

 

So, that's that...  I hope this makes sense.  

 

Alex



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Re: [asterisk-users] Detect if people is talking

[Sebastian Nielsen]

It sounds like there is more of the problem that neither the agent or customer 
knows when to start talking, ergo, when the call is "Connected", thus the OP 
wants the agent to start talking before the customer is brought in front of 
that agent.

Another solution would be to just play a "fake" recorded "hello" to both ends, 
maybe with a slight shift, inviting both to start talking.

I don't think its a problem with the agents failing to do their job, but rather 
unsuredness, maybe because it have happened regularly to those agents that they 
just "speak out in the empty" without any customer on the other end, and thus 
the agent instead waits for customer to say hello, while customer waits for 
agent to say hello.

Its a "classic problem" in the phone industry, so a great solution could be to 
play a fake hello after both of them are connected, inviting both to start 
talking, and they will automatically "find" each other.

-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com 
 För Steve Edwards
Skickat: den 31 december 2020 18:36
Till: Asterisk Users Mailing List - Non-Commercial Discussion 

Ämne: Re: [asterisk-users] Detect if people is talking

On Wed, 30 Dec 2020, Valter Nogueira wrote:

> We have some agents that pick calls but say nothing, letting customers 
> "alone". Is there any way to detect if an agent is speaking?

I'm not sure I understand the situation. Are you saying agents are failing to 
do their job and just let the customer wait until they hang up in frustration?

If you record the calls, could you analyze them after the call? I don't use 
agents or queues so I don't know if it is possible, but the 'monitor()' 
application records each leg in a separate file.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] Anyone that know of DECT "client" for asterisk?

[Sebastian Nielsen]

The problem is that there is only a few operators left (4 in sweden) of which 2 
have stopped accepting new customers at all.
And the 2 other requires a credit check with a minimum monthly wage of 1200$ 
(which is the "golden standard" for getting a loan) just because theres no 
"credit limit" and you could potentially call for as much as you want.

They aren't even interested in getting "low-credit" customers, and when I asked 
if they instead could put up a credit limit, they told me thats something they 
will do when the account is approved and opened - ergo its not a way of 
bypassing the wage limit, but is a way of own peace, and they also told me that 
the credit limit is not a gurantee that the bill will stay below the limit, it 
will only check during new calls, they will not terminate a call due to credit 
limit, which means a premium call could still overshoot the account.

Most operators in sweden use the TechniColor modem, which also is a DECT base 
station.
I think that such a solution would create a very reliable solution as long as 
the actual hardware is close to the DECT base station.

Since the DECT/GAP protocol is digital, theres no analog-digital or 
digital-analog conversion circuit needed.

I have seen a lot of DECT USB adapters, but these act "as a base station" (ergo 
accepts registrations from handsets) and not "as a handset".

If there is some hardware that is opposite, like "act as a handset" and could 
talk to a DECT base - it would be preferable.
Somebody that knows such hardware somewhere?
It must be someone that want to connect a asterisk server or other PC solution 
to a standard DECT base?


-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com 
 För Frank Vanoni
Skickat: den 7 oktober 2020 19:17
Till: asterisk-users@lists.digium.com
Ämne: Re: [asterisk-users] Anyone that know of DECT "client" for asterisk?

On Sat, 2020-10-03 at 22:25 +0200, Sebastian Nielsen wrote:

> many providers in sweden have started disabling SIP account details 
> and now require usage of their own ”router’s”.

That's very irritating and make me angry. Few of my client had the same 
problem. The solution: write a letter asking the SIP credentials explaining you 
want configure your own equipment and tell them you switch to another provider 
in case of refusal. Good luck!

I don't know if there is an appropriate hardware to build a DECT bridge and I 
doubt that fiddling with anything like that will not be a reliable solution.


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[asterisk-users] Anyone that know of DECT "client" for asterisk?

[Sebastian Nielsen]

Anyone here that knows some hardware USB "client" (DECT) that can connect to
a DECT base station and act as a "provider" in asterisk?

Ergo, asterisk is the DECT handset, and connects to the DECT base station.

Its also important that it also works with most asterisk-compatible devices,
including for example raspberry's.

 

The reason I ask, is that many providers in sweden have started disabling
SIP account details and now require usage of their own "router's". Using FXS
adapters is not an option as the digital->analog->digital conversion gives
very high echoes in the phones.

The idea is to have something simulate a DECT handset, connect to the
provider's router, and thus be able to still use asterisk.

 

Best regards, Sebastian Nielsen



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Re: [asterisk-users] Channels freeze on Confbridge

[Sebastian Nielsen]

>>I can see the point you're making here, but what's going to do this after 30
*minutes* of normal call?

I was more into, if there is some feature that somehow triggers after 30 
minutes of call - and this feature is unsupported on some client, which causes 
it to drop the call. For example, if you are trying to send some call cost 
notification for long calls out of band or similiar, and some devices doesn't 
support this feature.
 




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Re: [asterisk-users] Channels freeze on Confbridge

[Sebastian Nielsen]

I had a similiar problem, but with calls dropping after 30 sec.
It turned out that Android didn't support RP-CID (Reverse Party Caller ID) so 
when I sent the name of the callee to the caller (as some sort of "centralized 
phonebook function") it caused calls to be dropped as android refused to reply 
on the packets or sent rejections back.

Check if you have some equipment on the line which doesn't support a specific 
function, and configure the equipment to use a separate SIP account with these 
features turned off.

I first tought it would just ignore unsupported features, but it turned out it 
outright rejects packets with unsupported features.

Best regards, Sebastian Nielsen

-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com 
 För C.Maj
Skickat: den 22 augusti 2020 20:03
Till: asterisk-users@lists.digium.com
Ämne: Re: [asterisk-users] Channels freeze on Confbridge

On 2020-08-18 13:00, Carlos Chavez wrote:
> users complain that confbridge calls end after about 30 minutes or so

You might want to turn up SIP debug logging -- could be a re-INVITE is getting 
dropped, NAT pin-hole is closing, or some other network issue.

-- 
鸞 C. Maj, Technology Captain @ Penguin PBX Solutions
 USA Toll Free 1-833-PNGNPBX (1-833-764-6729)
酪 International & SMS Texting +1.720.32.42.72.9
 Visit on the World Wide Web at PENGUINPBX.COM

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Re: [asterisk-users] Stir-Shaken for asterisk

[Sebastian Nielsen]

Yes, this means that a provider which only provides IP-access (for example a 
broadband operator), ergo, when it doesn’t terminate a call, but where the call 
terminates directly at a enterprise, does not need to force the end customer to 
implement call verification in their PBX.
Basically, if you don’t have control of the SIP endpoint where the call is 
terminated, you don’t need to implement these rules.
 
Also this doesn’t apply to the customer end of the operator, where you 
authenticate to your operator with your username/password. These calls are 
already authenticated.
It applies to the so called ”anonymous” calls that traverses between operators 
and through operators networks.
 
If they don’t have access to the PBX equipment, and the owner is not required 
to be a FCC approved operator, then the rules are dropped.
SIP2SIP calls using textual URI’s are also not in scope for this rules, only 
DID calls are applicable.
 
Rule 1 also says for internal calls (ergo inside operator network) you need to 
implement a security solution CONSISTENT with stir/shaken, not in accordance.
It means you can roll your own solution, as long as it provides comparable 
security.
One example, is in call registry’s, limiting so customers can only use their 
own callerIDs as callerID.
 
I suspect that the reason FCC didn’t want to just implement callerID 
restrictions, is that they propably want to make it possible for US number 
owners, to use their numbers outside of the country. Else it would been easy to 
just force operators to restrict which numbers can be used inside phone 
networks, so international calls cannot have a US number as source, and calls 
inside USA must use their customer-assigned number as source, no other source.

Also the last rule about KYC means that anonymous pre-paid phone cards, both 
SIMs but also those scratch-off phone-cards with a access number, and also 
anonymous SIP accounts/DIDs will no longer be allowed, all calls must be able 
to be traced to either a corporation or a physical person.
 
Från: asterisk-users-boun...@lists.digium.com 
 För Jeff LaCoursiere
Skickat: den 28 maj 2020 06:11
Till: asterisk-users@lists.digium.com
Ämne: Re: [asterisk-users] Stir-Shaken for asterisk
 
A few weeks... like in a year and a few weeks:
https://transnexus.com/blog/2020/fcc-mandates-stir-shaken/
Some interesting bits in there as well, like:
"These rules do not apply to providers that lack control of the network 
infrastructure necessary to implement STIR/SHAKEN."
See also:
https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN
 

   
Jeff LaCoursiere
STRATUSTALK, INC. / CTO

Phone:
+1 703.496.4990 x108

Mobile:
+1 815.546.6599

Email:
  j...@stratustalk.com 

Website:
  https://www.stratustalk.com

Address:
One Freedom Square
13th Floor
Reston, VA 20190

 

   
On 5/27/20 10:51 PM, Saint Michael wrote:
In a few weeks, no SIP call is going to terminate unless they are signed 
properly, as mandated by law.  We are in the business of Stir-Shaken, signing 
calls, as an FCC-approved provider. A big differentiator between our service 
and the rest: we are the only ones who don't need to receive the calls in our 
servers to sign them. We do this over a MySQL call, easily connectable to 
Asterisk via res_odbc, so you never have to send us your calls. This is a 
sample of how we do this so you may test now:
mysql -u anonymous -h 208.73.232.47 -e "call 
strshk.stir_shaken_signature('7274433019','1957408')".
If your caller-ID is a valid US number and not a wireless number (that is a 
NO-NO for the FCC), we sign the call as 'C', if you use your own DIDs, 
something we can verify as legit, then we sign as 'B', and if you use our DID 
as caller ID, we sign as 'A', full attestation.  
Please email to venefax at g mail if you have any questions. Do not think you 
can do business as usual. The wild west of VOIP is coming to an end. But we can 
keep you in business if you follow the rules.





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Re: [asterisk-users] On Register, run a script, validate source IP

[Sebastian Nielsen]

You could use permit/deny in the sip.conf.

That would require your script to update sip.conf dynamically and reload the 
config for each time user wants to update their accepted location.

To avoid excessive reloads, you could have that the changes will take effect 
after 00:00, so you have a cron script which reads the user database and 
updates sip.conf, and then reloads asterisk ONCE.
So any changes user makes to their sourceIP/GeoIP configuration on webpage, 
will not take effect until midnight.

-Ursprungligt meddelande-
Från: asterisk-users  För Benoit 
Panizzon
Skickat: den 18 november 2019 13:23
Till: asterisk-users@lists.digium.com
Ämne: [asterisk-users] On Register, run a script, validate source IP

Hi Gang

To increase security against phished passwords and similar attacks, we consider 
offering customers to define IP ranges (or GeoIP locations) from which their 
dynamic registrations are being accepted.

I can already look at the source IP in the dial plan, so no issue with validate 
an INVITE against a source IP.

But I would also like to prevent registrations from outside of this client's 
specific allowed ip addresses as well, so the line cannot be hijacked.

So I'm looking for something like

On Register:
If check_allowed_ip(auth_username) {
return;
} else {
Reply(403 Wrong IP for this user);
}

Any ideas how to do that? (Yes, I asked Google and found nothing useful yet)

Mit freundlichen Grüssen

-Benoît Panizzon-
-- 
I m p r o W a r e   A G-Leiter Commerce Kunden
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CH-4133 PrattelnFax  +41 61 826 93 01
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Re: [asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for certain SIP peer

[Sebastian Nielsen]

FINALLY solved it… Googled around for the problem, and found this:

https://support.yeastar.com/hc/en-us/articles/360020908914-Call-Hangs-up-at-30-Seconds

 

Apparently, sendrpid=yes causes Android Native SIP client not to respond to the 
packets, and this drops the call after 30 seconds.

 

Disabling sendrpid makes it work successfully.

 

 

Från: asterisk-users  För Joshua C. 
Colp
Skickat: den 17 november 2019 01:18
Till: Asterisk Users Mailing List - Non-Commercial Discussion 

Ämne: Re: [asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for 
certain SIP peer

 

On Sat, Nov 16, 2019 at 7:59 PM Sebastian Nielsen mailto:sebast...@sebbe.eu> > wrote:

What would be the best way to solve this problem? Anyone else that have got the 
same problem with Android’s native SIP client, especially on Samsung phones?

 

I do not know if the bug is in Android native SIP, or Samsung’s build of the 
SIP client, or if the bug is even with the OpenVPN client, or where the bug 
actually is.

The ACK might even be sent for real, but have the incorrect source IP so 
asterisk ignores it.

 

The ACK is sent to the Contact header of the 200 OK sent to the phone. Using 
the respective logging (sip set debug on or pjsip set logger on) would tell you 
the IP address and port that Asterisk is telling the phone to send to, and 
isolate the problem further. Asterisk also doesn't ignore the ACK based on 
source IP address. If it shows up at Asterisk, it'll get processed.

 

 

Since audio works in both directions, it seems that the lack of ACK wouldn’t 
hurt (other than asterisk forcefully disconnecting the call) so I need to just 
tell Asterisk to not forcefully disconnect the callee.

 

Without modifying code there's no way. The 200 OK retransmits until it gives 
up, and the call is disconnected.

 

-- 

Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.sangoma.com <http://www.sangoma.com/>  & www.asterisk.org 
<http://www.asterisk.org/> 

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Re: [asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for certain SIP peer

[Sebastian Nielsen]

What would be the best way to solve this problem? Anyone else that have got the 
same problem with Android’s native SIP client, especially on Samsung phones?

 

I do not know if the bug is in Android native SIP, or Samsung’s build of the 
SIP client, or if the bug is even with the OpenVPN client, or where the bug 
actually is.

The ACK might even be sent for real, but have the incorrect source IP so 
asterisk ignores it.

 

Since audio works in both directions, it seems that the lack of ACK wouldn’t 
hurt (other than asterisk forcefully disconnecting the call) so I need to just 
tell Asterisk to not forcefully disconnect the callee.

 

Från: asterisk-users  För Joshua C. 
Colp
Skickat: den 17 november 2019 00:54
Till: Asterisk Users Mailing List - Non-Commercial Discussion 

Ämne: Re: [asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for 
certain SIP peer

 

On Sat, Nov 16, 2019 at 7:45 PM Sebastian Nielsen mailto:sebast...@sebbe.eu> > wrote:

Hello.

I have a problem with the native Android SIP client, not acknowledging the call.

 

Sent a message to the list for some weeks ago containing a sip debug log, but 
it only got stuck in moderation queue due to too large size (and it said I 
would get a message if moderators rejected it, but did not get message and I 
don’t think it got posted to list either)

 

This ONLY happens when calling outgoing from the Android SIP client. Incoming 
calls works flawlessly.

 

Everything works, audio in both directions, but the call is dropped after 30 
sec.

I have debugged it very much, and it seems that either Android is sending the 
acknowledge of the call to the incorrect IP (perhaps to the 3G network instead 
of via the VPN), or not sending it at all.

 

BUT – Everything else is working flawlessly, including audio in both directions.

 

So this means, I need somehow to tell Asterisk to ignore the lack of 
acknowledgement.

 

 

 

So now to the question, since the call is dropped automatically after 30 sec 
with ”NO_USER_RESPONSE” (Hangupcause 18) on the far end (the callee’s end), 
propably because the Android native Client is not acknowledging the connected 
call , is it possible to tell Asterisk to just ignore the lack of 
acknowledgement from Android somehow?

 

Basically, for Client sip09 (username), never hang up for the reason 18 
(NO_USER_RESPONSE), threat like user response was received always.

 

There is no ability to ignore the lack of an ACK, as that violates the SIP 
standard itself.

 

-- 

Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.sangoma.com <http://www.sangoma.com/>  & www.asterisk.org 
<http://www.asterisk.org/> 

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[asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for certain SIP peer

[Sebastian Nielsen]

Hello.

I have a problem with the native Android SIP client, not acknowledging the
call.

 

Sent a message to the list for some weeks ago containing a sip debug log,
but it only got stuck in moderation queue due to too large size (and it said
I would get a message if moderators rejected it, but did not get message and
I don't think it got posted to list either)

 

This ONLY happens when calling outgoing from the Android SIP client.
Incoming calls works flawlessly.

 

Everything works, audio in both directions, but the call is dropped after 30
sec.

I have debugged it very much, and it seems that either Android is sending
the acknowledge of the call to the incorrect IP (perhaps to the 3G network
instead of via the VPN), or not sending it at all.

 

BUT - Everything else is working flawlessly, including audio in both
directions.

 

So this means, I need somehow to tell Asterisk to ignore the lack of
acknowledgement.

 

 

 

So now to the question, since the call is dropped automatically after 30 sec
with "NO_USER_RESPONSE" (Hangupcause 18) on the far end (the callee's end),
propably because the Android native Client is not acknowledging the
connected call , is it possible to tell Asterisk to just ignore the lack of
acknowledgement from Android somehow?

 

Basically, for Client sip09 (username), never hang up for the reason 18
(NO_USER_RESPONSE), threat like user response was received always.

 

 

Best regards, Sebastian Nielsen

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[asterisk-users] Problems with calls dropping on Android.

[Sebastian Nielsen]

Hello.

I have the following in sip.conf

[sip09]

type=peer

defaultuser=sip09

nat=yes

qualify=no

secret=sip09

host=dynamic

context=outgoing

dtmfmode=rfc2833

disallow=all

allow=ulaw

allow=alaw

allow=h263p

deny=0.0.0.0/0.0.0.0

permit=192.168.2.2/255.255.255.255

jbenable = yes

jbforce = yes

jbmaxsize = 100

jbresyncthreshold = 200

jbimpl = fixed

transport=tcp

sendrpid=yes

 

And these settings in Android native client.

 

Username: sip09

Password: sip09

Server: 192.168.1.10

Username at authentication: sip09

Display name: Same as username

Outgoing proxy: 192.168.1.10

Port: 5060

Transport: TCP

Send keep alive: Always

 

However, if I make a call FROM android phone, call is dropped after 30
seconds, regardless of answer or not. If I make call TO android phone, it
works normally.

No NAT problems inbetween, there is a VPN between the phone and SIP server
with full access.

 

I guess I need to do some trick to have it work with Android. Apparently the
packets are received in both ends - else audio wouldn't work, but guess the
stock native SIP client on android ignores certain packets right?

This is an Android 9 phone.

 

 

Additionally, I wonder if its possible to change the callerid shown in
display when calling out? Like RPID. It works on my desktop phones, if I
enter a short code, the full name and number is shown on display, but on the
Android phone, it doesn't work, only the dialled shortnumber is shown.

Also I wonder if its possible to have asterisk send the remote callerid
(when receiving a call) in such a way it gets stored in call log with full
names and such - without having to resort to using phonebook.

 

 

SIP debug log:

 

*CLI> sip set debug ip 192.168.2.2

SIP Debugging Enabled for IP: 192.168.2.2

*CLI> Really destroying SIP dialog
'6f9956035553ab1b79ca057f5dffe0ac@192.168.2.2' Method: OPTIONS

Really destroying SIP dialog 'fc3307059c816094a6c6ce100cf383e5@192.168.2.2'
Method: OPTIONS

 

<--- SIP read from TCP:192.168.2.2:51729 --->

OPTIONS sip:192.168.1.10 SIP/2.0

Call-ID: e65234cb818a143bc3c167a782b98e96@192.168.2.2

CSeq: 3984 OPTIONS

From: "sip09" ;tag=3997716169

To: "sip09" 

Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK105b3648c13a72f8fbe7ce3049df71aa3130;rport

Max-Forwards: 70

User-Agent: SIPAUA/0.1.001

Content-Length: 0

 

<->

--- (9 headers 0 lines) ---

Sending to 192.168.2.2:51729 (no NAT)

Looking for s in cellip (domain 192.168.1.10)

 

<--- Transmitting (no NAT) to 192.168.2.2:51729 --->

SIP/2.0 200 OK

Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK105b3648c13a72f8fbe7ce3049df71aa3130;receive
d=192.168.2.2;rport=51729

From: "sip09" ;tag=3997716169

To: "sip09" ;tag=as4c9bb00e

Call-ID: e65234cb818a143bc3c167a782b98e96@192.168.2.2

CSeq: 3984 OPTIONS

Server: Asterisk PBX 13.21.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE

Supported: replaces,timer

Contact: 

Accept: application/sdp

Content-Length: 0

 

 

<>

Scheduling destruction of SIP dialog
'e65234cb818a143bc3c167a782b98e96@192.168.2.2' in 32000 ms (Method: OPTIONS)

 

<--- SIP read from TCP:192.168.2.2:51729 --->

INVITE sip:02@192.168.1.10 SIP/2.0

Call-ID: fcaad738faee2d0250d0cf2366139979@192.168.2.2

CSeq: 9116 INVITE

From: "sip09" ;tag=3432177901

To: 

Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bKf8bf8138c906000bc3f8601a5df558943130;rport

Max-Forwards: 70

Contact: "sip09" 

Content-Type: application/sdp

Content-Length: 295

 

v=0

o=- 1571035683065 1571035683066 IN IP4 192.168.2.2

s=-

c=IN IP4 192.168.2.2

t=0 0

m=audio 26726 RTP/AVP 96 97 3 0 8 127

a=rtpmap:96 GSM-EFR/8000

a=rtpmap:97 AMR/8000

a=rtpmap:3 GSM/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:127 telephone-event/8000

a=fmtp:127 0-15

<->

--- (10 headers 13 lines) ---

Sending to 192.168.2.2:51729 (no NAT)

Sending to 192.168.2.2:51729 (no NAT)

Using INVITE request as basis request -
fcaad738faee2d0250d0cf2366139979@192.168.2.2

Found peer 'sip09' for 'sip09' from 192.168.2.2:51729

 

<--- Reliably Transmitting (NAT) to 192.168.2.2:51729 --->

SIP/2.0 401 Unauthorized

Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bKf8bf8138c906000bc3f8601a5df558943130;receive
d=192.168.2.2;rport=51729

From: "sip09" ;tag=3432177901

To: ;tag=as4d53b5f5

Call-ID: fcaad738faee2d0250d0cf2366139979@192.168.2.2

CSeq: 9116 INVITE

Server: Asterisk PBX 13.21.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE

Supported: replaces,timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6dc98e50"

Content-Length: 0

 

 

<>

Scheduling destruction of SIP dialog
'fcaad738faee2d0250d0cf2366139979@192.168.2.2' in 32000 ms (Method: INVITE)

 

<--- SIP read from TCP:192.168.2.2:51729 --->

ACK sip:02@192.168.1.10 SIP/2.0

Call-ID: fcaad738faee2d0250d0cf2366139979@192.168.2.2

Max-Forwards: 70

From: "sip09" ;tag=3432177901

To: 

Re: [asterisk-users] Paging systems?

[Sebastian Nielsen]

The thing is:

Does the paging system connect to a line (like it was a deskphone) or does
the paging system ACT as a line (that you connect a deskphone to)?

 

If the page system is technically a phone, the it should work with a SPA.
What you need to do then, is to figure out what the line does to get the
paging system to auto-answer.

The first thing you could do, is to connect a regular home-phone to the same
jack that the paging system were PREVIOUSLY connected, and then try ”paging”
it.

 

Then check the display. It could display a specific caller ID (that you need
to fake inside Asterisk) or it could send specific signals (which you hear
on the rings).

If you then send this ”fake” callerid from the asterisk to the SPA, it will
also send out this ”fake” callerid out to the paging system and cause it to
answer.

 

 

Or it could be the opposite, the paging system IS the line, and you
technically connect a line-out port to the paging system, ergo, the page
system acts like a phone company’s line in the wall.

Then you need something with a FXS port (something that acts like a phone).

 

 

 

Från: asterisk-users  För Michael
Munger
Skickat: den 21 mars 2019 21:05
Till: asterisk-users@lists.digium.com
Ämne: Re: [asterisk-users] Paging systems?

 

It worked on the old system.

I am open to suggestions, but don't want (or have the option) to add a TDM
card.

 





Michael Munger, dCAP, MCPS, MCNPS, MBSS


Microsoft Certified Professional


Microsoft Certified Small Business Specialist


Digium Certified Asterisk Professional


High Powered Help, Inc.


p:

678-905-8569


w:

 <https://hph.io> hph.io  e:  <mailto:m...@hph.io> m...@hph.io






On 3/21/19 3:01 PM, Sebastian Nielsen wrote:

How did the page system answer the call when it was used with the analog
system?

You could propably ”fake” those signals from inside asterisk, and cause it
to answer.

 

Från: asterisk-users  <mailto:asterisk-users-boun...@lists.digium.com>
 För Michael Munger
Skickat: den 21 mars 2019 20:00
Till: asterisk-users@lists.digium.com
<mailto:asterisk-users@lists.digium.com> 
Ämne: [asterisk-users] Paging systems?

 

Does anyone have an (overhead) paging system that they like that works with
SIP?

 

We’ve got a client with an old paging system that (supposedly) just takes an
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t
auto-answer the call, so paging never happens.

 

 





Michael J. Munger, dCAP, MCPS, MCNPS, MBSS


Microsoft Certified Professional


Microsoft Certified Small Business Specialist


Digium Certified Asterisk Professional


High Powered Help, Inc.


p:

678-905-8569


w:

 <https://hph.io> hph.io  e:  <mailto:m...@hph.io> m...@hph.io

 

 







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Re: [asterisk-users] Paging systems?

[Sebastian Nielsen]

How did the page system answer the call when it was used with the analog
system?

You could propably ”fake” those signals from inside asterisk, and cause it
to answer.

 

Från: asterisk-users  För Michael
Munger
Skickat: den 21 mars 2019 20:00
Till: asterisk-users@lists.digium.com
Ämne: [asterisk-users] Paging systems?

 

Does anyone have an (overhead) paging system that they like that works with
SIP?

 

We’ve got a client with an old paging system that (supposedly) just takes an
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t
auto-answer the call, so paging never happens.

 

 





Michael J. Munger, dCAP, MCPS, MCNPS, MBSS


Microsoft Certified Professional


Microsoft Certified Small Business Specialist


Digium Certified Asterisk Professional


High Powered Help, Inc.


p:

678-905-8569


w:

  hph.io  e:   m...@hph.io

 

 



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Re: [asterisk-users] Queue not dialing out to cell phone for some reason

[Sebastian Nielsen]

Aha, I tought you had a SIP client (like MizuDroid or similiar) that registred 
via data connection to the asterisk server.

 

Seems theres a problem with the trunk then.

 

What does ”sip show registry” tell you?

(asterisk -r in console and then sip show registry)

 

It should show a status of ”Registred” to your trunk operator.

 

Från: Ivan Demkovitch  
Skickat: den 15 november 2018 18:01
Till: Sebastian Nielsen ; 'Asterisk Users Mailing List - 
Non-Commercial Discussion' 
Ämne: Re: SV: [asterisk-users] Queue not dialing out to cell phone for some 
reason

 

Sebastian,

 

I don't think it has to do anything with registration. It is dialing through 
the SIP trunk, so it goes out as normal cell phone call.

Also, why I don't see anything in a log? I see only first 2 members being 
dialed. 

 

  _  

From: Sebastian Nielsen mailto:sebast...@sebbe.eu> >
To: 'Ivan Demkovitch' mailto:idemkovi...@yahoo.com> >; 
'Asterisk Users Mailing List - Non-Commercial Discussion' 
mailto:asterisk-users@lists.digium.com> > 
Sent: Thursday, November 15, 2018 10:58 AM
Subject: SV: [asterisk-users] Queue not dialing out to cell phone for some 
reason

 

I would suspect that the cell phone does use battery saving causing the SIP 
application to lose registration with the server. Would also suggest using TCP 
with a fairly short keepalive to prevent the cellular network from tearing down 
the connection to the asterisk server.

You need to go into android settings and make sure the SIP client is 
whitelisted in battery management.

 

Från: asterisk-users mailto:asterisk-users-boun...@lists.digium.com> > För Ivan Demkovitch
Skickat: den 15 november 2018 17:55
Till: asterisk-users@lists.digium.com <mailto:asterisk-users@lists.digium.com> 
Ämne: [asterisk-users] Queue not dialing out to cell phone for some reason

 

Hello,

 

I have queues.conf setup with a group like so:

 

[Sales](StandardQueue)
announce = first
member => SIP/FF4C119EEBF8-SLS
member => SIP/FF9EF375CCFC-SLS
member => SIP/1314555@callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink

 

So, my idea here that it should ring all 4 phones at the same time. And it does 
work but randomly.

I did trace a call and this is what I see. Only 2 phones (internal) called. 
External SIP@callcentric is not being called.

 

Any idea why it's not being called?

 


-- Executing [1@automated_attendant_normal:1] 
Verbose("SIP/callcentric15-0435", "1, Caller "DEMKOVITCH,IVAN" 
<13144880983> has entered the sales queue") in new stack
  Caller "aa" <155> has entered the sales queue
-- Executing [1@automated_attendant_normal:2] 
Goto("SIP/callcentric15-0435", "queues,7001,1") in new stack
-- Goto (queues,7001,1)
-- Executing [7001@queues:1] Verbose("SIP/callcentric15-0435", "2,"aa" 
<155> entering sales queue") in new stack
  == "aa" <155> entering sales queue
-- Executing [7001@queues:2] BackGround("SIP/callcentric15-0435", 
"/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
--  Playing 
'/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')
-- Executing [7001@queues:3] Queue("SIP/callcentric15-0435", 
"sales85") in new stack
-- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0437 is ringing
-- SIP/FF9EF375CCFC-SLS-0436 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
--  Playing 'queue-periodic-announce.ulaw' 
(language 'en')
-- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0439 is ringing
-- SIP/FF9EF375CCFC-SLS-0438 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
--  Playing 'queue-periodic-announce.ulaw' 
(language 'en')
-- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-043b is ringing
-- SIP/FF9EF375CCFC-SLS-043a is ringing
-- Stopped music on hold on SIP/callcentric15-0435
  == Spawn extension (queues, 7001, 3) exited non-zero on 
'SIP/callce

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

[Sebastian Nielsen]

I would suspect that the cell phone does use battery saving causing the SIP 
application to lose registration with the server. Would also suggest using TCP 
with a fairly short keepalive to prevent the cellular network from tearing down 
the connection to the asterisk server.

You need to go into android settings and make sure the SIP client is 
whitelisted in battery management.

 

Från: asterisk-users  För Ivan 
Demkovitch
Skickat: den 15 november 2018 17:55
Till: asterisk-users@lists.digium.com
Ämne: [asterisk-users] Queue not dialing out to cell phone for some reason

 

Hello,

 

I have queues.conf setup with a group like so:

 

[Sales](StandardQueue)
announce = first
member => SIP/FF4C119EEBF8-SLS
member => SIP/FF9EF375CCFC-SLS
member => SIP/1314555@callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink

 

So, my idea here that it should ring all 4 phones at the same time. And it does 
work but randomly.

I did trace a call and this is what I see. Only 2 phones (internal) called. 
External SIP@callcentric is not being called.

 

Any idea why it's not being called?

 


-- Executing [1@automated_attendant_normal:1] 
Verbose("SIP/callcentric15-0435", "1, Caller "DEMKOVITCH,IVAN" 
<13144880983> has entered the sales queue") in new stack
  Caller "aa" <155> has entered the sales queue
-- Executing [1@automated_attendant_normal:2] 
Goto("SIP/callcentric15-0435", "queues,7001,1") in new stack
-- Goto (queues,7001,1)
-- Executing [7001@queues:1] Verbose("SIP/callcentric15-0435", "2,"aa" 
<155> entering sales queue") in new stack
  == "aa" <155> entering sales queue
-- Executing [7001@queues:2] BackGround("SIP/callcentric15-0435", 
"/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
--  Playing 
'/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')
-- Executing [7001@queues:3] Queue("SIP/callcentric15-0435", 
"sales85") in new stack
-- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0437 is ringing
-- SIP/FF9EF375CCFC-SLS-0436 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
--  Playing 'queue-periodic-announce.ulaw' 
(language 'en')
-- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0439 is ringing
-- SIP/FF9EF375CCFC-SLS-0438 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
--  Playing 'queue-periodic-announce.ulaw' 
(language 'en')
-- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-043b is ringing
-- SIP/FF9EF375CCFC-SLS-043a is ringing
-- Stopped music on hold on SIP/callcentric15-0435
  == Spawn extension (queues, 7001, 3) exited non-zero on 
'SIP/callcentric15-0435'



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[asterisk-users] Detect missed call in extensions?

[Sebastian Nielsen]

How I do to detect missed calls?

 

After Dial() has been executed, theres 3 ways a call could end up in:

 

1: The callee answers, and a communication is going on. Then one party hangs
up, and thus execution goes to the h extension.

2: The callee newer answers or there was some error, the Dial() fails, and
execution continues on next line in extensions.

3: The caller hangs up before callee have answered, and execution goes to
the h extension.

 

Now to the problem. I want to detect if callee did answer or not (in
separate 1 and 3) so I could determite if a missed call should be logged to
a missedcall.txt log file. (call should be logged in 3 case, but not in 1
case)

2 is easy to detect, as these always are failed (non-answered) calls.

 

Best regards, Sebastian Nielsen

 



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Re: [asterisk-users] Callee id over chan_sip trunk

[Sebastian Nielsen]

I found very useful info here:
https://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE

In other words, on the asterisk1 box, you need to fetch from SIPPEER in
extensions on asterisk1 box, and then populate connectedline.
SIPPEER is the callee leg of the call, and CONNECTEDLINE is the caller. So
if you set CONNECTEDLINE on caller (eg the asterisk2 side of the trunk
between asterisk1 and asterisk2),
You need to fetch this info in extensions for the SIPPEER on asterisk1 side
of the trunk between asterisk1 and asterisk2, and copy this info into
CONNECTEDLINE (the ISDN PRI leg of the call) on the asterisk1 box.

I guess you have a extension on asterisk2, and then call "through" asterisk1
box.

(Otherwise, if you are "behind" asterisk2 box and call the Conf line on
asterisk1, you need to do the opposite of above, set the things on asterisk2
box.)

-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] För Dmitry Melekhov
Skickat: den 15 maj 2017 12:47
Till: Asterisk Users Mailing List - Non-Commercial Discussion

Ämne: [asterisk-users] Callee id over chan_sip trunk

Hello!


I run two asterisks 13.13.1.


Here is how they are connected:


me---PBX--isdn pri--asterisk1--sip--asterisk2.


If I call something from asterisk1 and have in dial plan:

Let's say

exten => 6000,n,Set(CONNECTEDLINE(name)=Conf. 6000)

exten => 6000,n,Meetme(6000,TL(1080:6))


Then I see Conf. 6000 on my phone if I call 6000.


If I have the same code for number on asterisk2, then there is no name on my
phone,

i.e. looks like asterisk doesn't send this info, at least I don't see it in
sip debug.


Could you tell me is it possible to pass this over sip?

Thank you!





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Re: [asterisk-users] How to detect fake CallerID? (8xx?)

[Sebastian Nielsen]

Personally, if I was a client, I would rather have the personell answer the
phone than make a outgoing call, if I would choose.
If you think of billing and costs.
So if a client allows outgoing, I don't think they have any problems with
answering a call immediately following either.

But I assume the client will be billed for the time the personell works
there?
And thats why you have this "phone verification system", to avoid discussion
about how long the company has been there and unfair bills?

Then you could have it this way instead:
1: Give the client (not personell) a PIN code.
2: The client calls and enters PIN.
3: The employee gets a SMS/email/push message/paging tone, that he can start
working.
4: When the employee is done, the client calls again, and enter PIN. This
will stop billing.
5: When billing is stopped, the employee gets a SMS/email/push
message/paging tone he can stop working.


This will be rock solid. The employee only needs to check for the SMSes.
The SMSes prevent the client from cheating the system to get cheaper
service, like claiming to start when client do not, or calling for stop
before the employee is finished, because the employee will only work when he
get start signal, and will stop working at stop signal.

Theres no risk that the client will call in and check in/check out when the
employee is not there, because that would cause the client to
Be billed for rendered services.


-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] För Don Kelly
Skickat: den 11 maj 2017 17:04
Till: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Ämne: Re: [asterisk-users] How to detect fake CallerID? (8xx?)

As a client, I don't want service company personnel answering my phone.

As a service company, I don't want my clients thinking that I do not trust
my employees who are at the client facility.

  --Don


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Goldberg
Sent: Thursday, May 11, 2017 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to detect fake CallerID? (8xx?)

Seems like this is the best idea (challenge-response), a callback.  No
matter the callerid, you don't know where the caller is.  But if you place a
call BACK to the callerid, it's going to go to the destination.  Then you
either need the phone to be answered, or the phone to be answered and and
the challenge entered.


Adam Goldberg
AGP, LLC
+1-202-507-9900

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J Montoya or A
J Stiles
Sent: Thursday, May 11, 2017 7:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] How to detect fake CallerID? (8xx?)

On Wednesday 10 May 2017, Steve Edwards wrote:
> On Wed, 10 May 2017, J Montoya or A J Stiles wrote:
> > Presumably your staff carry mobile phones.  What about an app that 
> > gets the ID of the cell tower to which it is connected, and passes 
> > it and the SIM number in a HTTP request to a server you control?
> 
> The problem is that they are supposed to use the 'site landline' to 
> confirm presence -- not their cell phone with the spoofed CID.

Yes; but the whole point is that the caller ID from the site landline is no
longer reliable enough as evidence, by itself, that somebody is actually
there.

A custom app could read the ID of the cell tower to which it was connected
-- or even the phone's GPS co-ordinates -- and transmit that back to base
over the Internet.  Preferrably with some sort of precautions to make the
request harder to forge  (i.e., *not* just a plain HTTP GET with the MCC,
MNC, LAC and CID in the query string).  If your app makes its connection via
the site's wi- fi  (which will require the co-operation of the client)  as
opposed to the mobile network, so much the better, as there will be an IP
address against which you can match.


If you insist to use the site landline for your authentication, you could
extend the protocol to a full challenge-and-response as follows:  Play a
series of digits down the line to the caller, return the call as soon as
they hang up, and ask them to dial the same digits they just heard.  All
this can be done in the dialplan  (you might need to record some
announcements of your own, such as "Please memorise the following digits"
and "Please dial the digits you heard in the last call").  

Intercepting incoming calls *to* a number is much harder  (usually requiring
the co-operation of telcos, unless the interloper has access to some
equipment through which they know that the call will be routed; that
potentially includes your Asterisk, but any tampering there would be
evident)  than falsifying outgoing calls 

Re: [asterisk-users] How to detect fake CallerID? (8xx?)

[Sebastian Nielsen]

Since the callback happens immediately after hangning up, the risk of
answering a call that isn't theirs is minimal.
For those sites that divert their incoming calls to a PBX or answering
machine, you could have some config/database that excepts these sites from
callback verification.
(which means these sites run into risk of fake callerID).


Another variant could be that they must visit a specific website using a
Wifi or computer at the client. You record the IP.
Spoofing the IP in a TCP three-way handshake is almost impossible.

The thing is then to be able to record which IP is the client, but if your
services are ordered by the client via some web form, you could have that IP
be recorded as "client IP" and the employee must check in/check out from
that IP.

This could be used in unison with the phone verification, so the employee
can select which fits best for the enviroment.
(eg, they choose phone verification or web verification)

-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] För Don Kelly
Skickat: den 10 maj 2017 22:08
Till: 'Asterisk Users Mailing List - Non-Commercial Discussion'
<asterisk-users@lists.digium.com>
Ämne: Re: [asterisk-users] How to detect fake CallerID? (8xx?)

It's probably not practical to have them answering the client's telephone!
At a lot of sites, incoming calls would be handled by auto attendant,
diverted to answering service, etc.

  --Don


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
Nielsen
Sent: Wednesday, May 10, 2017 2:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to detect fake CallerID? (8xx?)

Use a callback.
So when clocking in/out, they will hear a random 4 digit PIN, like "Enter
four, three, six, eight at the callback".
After they hangup, the phone will ring, and then they will have confirm with
the 4 digit PIN.

If they arent in presence: the phone at the site will ring, and the person
at site (that isn't your employee) cannot carelessly just OK it because they
haven't heard the PIN.
If they are in presence: the phone at the site will ring, and the employee
will be able to enter the PIN they just heard. If they fake the callerID or
not at the initial call, does not matter, since you have verified with a
callback.

-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] För Steve Edwards
Skickat: den 10 maj 2017 19:13
Till: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Ämne: Re: [asterisk-users] How to detect fake CallerID? (8xx?)

On Wed, 10 May 2017, J Montoya or A J Stiles wrote:

> Presumably your staff carry mobile phones.  What about an app that 
> gets the ID of the cell tower to which it is connected, and passes it 
> and the SIM number in a HTTP request to a server you control?

The problem is that they are supposed to use the 'site landline' to confirm
presence -- not their cell phone with the spoofed CID.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] How to detect fake CallerID? (8xx?)

[Sebastian Nielsen]

Use a callback.
So when clocking in/out, they will hear a random 4 digit PIN, like "Enter
four, three, six, eight at the callback".
After they hangup, the phone will ring, and then they will have confirm with
the 4 digit PIN.

If they arent in presence: the phone at the site will ring, and the person
at site (that isn't your employee) cannot carelessly just OK it because they
haven't heard the PIN.
If they are in presence: the phone at the site will ring, and the employee
will be able to enter the PIN they just heard. If they fake the callerID or
not at the initial call, does not matter, since you have verified with a
callback.

-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] För Steve Edwards
Skickat: den 10 maj 2017 19:13
Till: Asterisk Users Mailing List - Non-Commercial Discussion

Ämne: Re: [asterisk-users] How to detect fake CallerID? (8xx?)

On Wed, 10 May 2017, J Montoya or A J Stiles wrote:

> Presumably your staff carry mobile phones.  What about an app that 
> gets the ID of the cell tower to which it is connected, and passes it 
> and the SIM number in a HTTP request to a server you control?

The problem is that they are supposed to use the 'site landline' to confirm
presence -- not their cell phone with the spoofed CID.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

[Sebastian Nielsen]

You need to ensure that traffic to the SIP box is sent to the correct IP. Also 
if you use split-tunnel (eg: not redirect-gateway def1) you must make sure NAT 
and traffic redirection works as is so the Asus router knows it should send the 
traffic through tunnel and not via WAN.
IMPORTANT: Then you must, in the ASUS RT-N66U make a port forward inwards from 
TUN to the phone client.
I would suggest wiresharking on the client side and see which IP Asterisk 
suggest the client should connect back to. This should be the internal IP of 
the asterisk server as seen from the openvpn server's point of view.
Another important thing: The local network in the Openvpns machine locatiin, 
may NOT have same subnet as the network behind the asus.All these must be 
separate, like:server network: 192.168.1.0/24Openvpn tunnel network: 
192.168.2.0/24Asus network: 192.168.3.0/24
Else you get bizarre routing problems when states appear in the state table.
 Originalmeddelande Från: Ernie Dunbar  
Datum: 2017-04-19  00:25  (GMT+01:00) Till: 'Asterisk Users Mailing List - 
Non-Commercial Discussion'  Rubrik: 
[asterisk-users] SIP connections over OpenVPN connection getone-way voice. 

Hi everyone. I'm having some trouble with an OpenVPN tunnel that
isn't working *quite* as well as we'd hoped.



First, here's our technical details:



The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a NAT
router. The router has UDP port 1194 forwarded to our server. This
server also runs our office Asterisk PBX, so there isn't any
networking hardware or firewall between the VPN tunnel and the
Asterisk PBX.



The OpenVPN client is an Asus RT-N66U router, which if I'm not
mistaken, runs a somewhat modified version of Tomato. 



I've got the VPN tunnel working well enough. I can do practically
anything from a computer hooked up to the client router as if I were
in the main office where the server is. But any SIP client I use -
whether it's a hardware SIP phone or a soft phone like Zoiper, can
connect to the Asterisk server without issue. Making calls can work,
accepting calls works, but I only get 1 way voice traffic. I can
hear voice data coming in FROM the Asterisk PBX, but I cannot send
any. 



In my experience with SIP, this usually means a firewall is breaking
the connection from the client phone to the Asterisk server. I just
can't for the life of me find what could be wrong. None of the other
traffic is being blocked. The ipfw firewall on the Asterisk PBX is
extremely open (see below). The firewall on the client router is
turned off, and as far as I can tell, most NAT routers don't even
block outbound traffic in the first place.



I can't see how traffic from the TUN interface on the OpenVPN server
even can be blocked going to another IP address on the same box, but
here are the IPFW rules:



root@ldinfo:/etc/asterisk# iptables -L -n

Chain INPUT (policy ACCEPT)

target prot opt source destination

ACCEPT all -- 192.168.0.0/24 192.168.0.3

ACCEPT all -- 192.168.1.0/24 192.168.0.3

ACCEPT all -- 10.8.0.0/24 192.168.0.3

ACCEPT all -- X.X.X.X 192.168.0.3

ACCEPT all -- 192.168.0.3 X.X.X.X

ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpt:1194

REJECT all -- 112.220.127.26 0.0.0.0/0 reject-with
icmp-port-unreachable



Chain FORWARD (policy ACCEPT)

target prot opt source destination



Chain OUTPUT (policy ACCEPT)

target prot opt source destination



Chain POSTROUTING (0 references)

target prot opt source destination



192.168.0.0/24 is the network the Asterisk PBX and OpenVPN server
are on.

192.168.1.0/24 is the network that the remote router is on.

10.8.0.0/24 is the network that the TUN device creates.

X.X.X.X is our datacenter.

192.168.0.3 is the IP address of our PBX.



Any assistance would be greatly appreciated.




  

  

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Re: [asterisk-users] semi-OFF-TOPIC - SIP iptables and NAT - same source, different destination

[Sebastian Nielsen]

Yes its called the state table. This because connection IP:PORT has a 
relationship with inside IP 192.168.x.x port X.

 

I guess you have configured the redirect port to be same on both?

Eg 5070 goes to *1:5060 and 5080 goes to *2:5060

 

What you need to do, is to have different inside ports as well, and also 
configure the asterisk boxes to listen on a different SIP port.

 

Från: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] För Gabriel Ortiz Lour
Skickat: den 27 januari 2017 19:59
Till: Asterisk Users Mailing List - Non-Commercial Discussion 

Ämne: [asterisk-users] semi-OFF-TOPIC - SIP iptables and NAT - same source, 
different destination

 

Hi all,

  anyone with iptables master power pack knowledge :) ?

 

  Having some problem with NAT!

  I have a server that is the LAN gateway (A) with the public IP, and two 
asterisk boxes behind it.

 

  I've configured port forward so port 5070 goes to *1 and 5080 goes to *2. 
Working fine.

 

  The problem is when some machine outside tries to talk with both asterisks.

  As soon as the 1st package gets routed to *1 the subsequent packets will all 
also get routed to *1, no matter that the destination port is now 5080.

 

  Seams like some "nat cache", where it will decide to forward all packets to 
*1 that come from origin "IP:PORT" X (since it was the first one contacted)

 

  anyone with iptables master power pack knowledge :) ?

Att.

Gabriel



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Re: [asterisk-users] Asterisk compatibility with SMS services

[Sebastian Nielsen]

Im using SMS successfully over VoIP. No problems at all. You however need to 
use a good codec.

 

However, I don’t use the MessageSend application, instead I use the raw SMS() 
application.

This works by the SMS centre calling my fixed landline from a specific number, 
I detect the callerid, initiate a SMS reception and then the SMS is in the 
spool files.

If I want to send a outgoing SMS, I push a SMS file in the spool folder, then 
initate a call to the SMS centre.

 

Ergo, incoming is like this (as early as possible in the dialplan):

 

exten => s,[any],GotoIf($[${CALLERID(num)} = 0740940]?recvsms,s,1)

(where 0740940 is your SMS centre number)

 

Then at recvsms:

[recvsms]

exten => s,1,SMS(in-${clid},a)

exten => s,2,System(/usr/sbin/mailbot sms ${clid})

exten => s,3,Hangup()

 

(where /usr/sbin/mailbot is a script that reads /var/spool/asterisk/mtrx 
folder, and clid is a variable containing a random number)

 

 

 

Outgoing is like this:

Place a file named [RandomA].[RandomB] into /var/spool/asterisk/motx with the 
following content:

da=[number you want to send to]

ud=[text you want to send, only GSM alphabet supported, composite messages NOT 
supported]

 

Then create the following file into /var/spool/asterisk/tmp:

Channel: SIP/074094@YOUROPERATOR

Callerid: "[YOURNUM]" <[YOURNUM]>

Application: SMS

Data: [RandomA]

 

(where 074094 is your SMS center)

Move the file from tmp into /var/spool/asterisk/outgoing/ and the SMS will be 
sent.

 

 

Från: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] För Brandon B.
Skickat: den 29 november 2016 17:25
Till: asterisk-users@lists.digium.com
Ämne: [asterisk-users] Asterisk compatibility with SMS services

 

Can anyone comment on using SMS in conjunction with VoIP service using one of 
these three VoIP providers: voip.ms, vitelity.com, flowroute.com? Are some SMS 
services more compatible with Asterisk (i.e. SMS over SIP works perfectly or 
not)? Is it best to use a different data channel for SMS messages (i.e. SMS via 
HTTP, SMS via XMPP) instead of Asterisk's built in SMS application MessageSend 
 ? In order to develop a 
web application for sending and receives SMS messages for business users, are 
there any pitfalls in using Asterisk to handle the message exchanges?



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Re: [asterisk-users] RS485 Audio device

[Sebastian Nielsen]

Why RS485? Whats wrong with a simple 3-wire connection (monospeaker, monomic, 
ground) where you short monomic to ground on button press?

Then you could use a simple usb device + device server to convert fron 
"smartphone headset" to usb then to network.

On the server, you use a SIP phone client, who use this device as mic/speaker, 
which is configured to lift the hook on headset button press.
In asterisk dialplan, you have logic which automatically dials where the 
doorphone should call upon hooklift.

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[asterisk-users] Problems with REGEXP - anchor string to beginning

[Sebastian Nielsen]

In extensions, I have this.

The variable "oex" contains the original extension called, and is used to
route outgoing calls internal or external depending on several factors.

 

But now, im implementing a system that should require a passcode upon
calling a "sensitive number".

 

Here is the relevant part in extensions.conf:

exten =>
s,12,Set(barrfile=${FILE(/var/secure_files/callbarring.txt,0,1,l,u)})

exten => s,13,Set(passcode=${SHIFT(barrfile)})

exten => s,14,Set(barrnumbers=${SHIFT(barrfile)})

exten => s,15,GotoIf($[${REGEX("${barrnumbers}",${oex})} = 0]?outgoing,s,17)

exten => s,16,Authenticate(${passcode})

; Testing purposes, to not route test calls out on the PSTN.

exten => s,17,Playback(you-have-reached-a-test-number)

exten => s,18,Hangup()

; Comment the above 2 lines and uncomment the next, to enable live PSTN
operation.

;exten => s,17,Dial(SIP/${oex}@cellip)

 

callbarring.txt contains (passcode is changed in this example, I just used
 as example):

,^11|020|9|09|00,

 

The thing is, that if I call a number that CONTAINS for example 09, it will
ask for passcode, but it should only ask for passcode if it BEGINS (note the
^) on that number.

So I have tested with:

,^(11|020|9|09|00),

And

,^11|^020|^9|^09|^00,

 

But with the result that the REGEX does never match and I get to the "You
have reached a test number" without authenticating, even if the number
begins on 020.

 

 

How I do to anchor to the beginning of the string in REGEX?

 



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

[Sebastian Nielsen]

Theres always garbage in the end of the files.

 

I do this when I want to read a file:

same => n,Set(featurefile=/home/test/feature-1.txt)
same => n,Set(unfilteredfeat2=${FILE(${featurefile},0,1,l,u)})

same => n,Set(feature2=${SHIFT(unfilteredfeat2)})

 

After that, add a , inside end of the file, so

 

Cat feature-1.txt

Reads:

radio,

 

Thus if there is garbage in the file, it will happen after the ,. Same if 
garbage happens to find its way into the end of the variable for some reason.





Från: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] För Jonathan H
Skickat: den 17 oktober 2016 15:32
Till: Asterisk Users Mailing List - Non-Commercial Discussion 

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

 

I have a plain text file, ASCII, unix line breaks. 1 single line, and all that 
is in it is the word "radio".

Here's some test dialplan:

exten => 5,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN})

same => n,Set(feature=${FILE(/home/test/feature-1.txt,0,1,l,u)})
same => n,Verbose(${feature})
   
same => n,Set(featurefile=/home/test/feature-1.txt)
same => n,Set(feature2=${FILE(${featurefile},0,1,l,u)})
same => n,Verbose(${feature2})

Both should output "radio", right? Here's the output:

-- Executing [5@fromvoipfone201:2] Set("PJSIP/6001-0052", 
"feature=radio") in new stack
-- Executing [5@fromvoipfone201:3] Verbose("PJSIP/6001-0052", "radio") 
in new stack
radio

-- Executing [5@fromvoipfone201:4] Set("PJSIP/6001-0052", 
"featurefile=/home/test/feature-1.txt") in new stack
-- Executing [5@fromvoipfone201:5] Set("PJSIP/6001-0052", 
"feature2=radi") in new stack 

GARRRGGG!  ^

 

-- Executing [5@fromvoipfone201:6] Verbose("PJSIP/6001-0052", "radi") 
in new stack

And this is what's just at the top of the script. If I put it way down in 
another context:

-- Executing [s@track-handler:3] Verbose("Local/s@root-0026;2", 
"/home/test/feature-1") in new stack
/home/test/feature-1
-- Executing [s@track-handler:4] Set("Local/s@root-0026;2", 
"feature=radio▒▒") in new stack 

EVEN MORE GARRRGGG! 
 ^


[Oct 17 13:29:33] ERROR[5093][C-003c]: json.c:704 ast_json_vpack: Error 
building JSON from '{s: s, s: s}': Invalid UTF-8 string.
[Oct 17 13:29:33] ERROR[5093][C-003c]: stasis_channels.c:773 
ast_channel_publish_varset: Error creating message
-- Executing [s@track-handler:5] Verbose("Local/s@root-0026;2", 
"radio▒▒") in new stack
radio▒▒
-- Executing [s@track-handler:6] GotoIf("Local/s@root-0026;2", 
"1?radio▒▒,s,1") in new stack

 

So, at the top of the file, when it's just a straight file as the filename, it 
gives the full word. If the filename is a variable, it strips and character. 
And further down, same thing, but it adds two weird blocks, which appear side 
by side in the console, but weirdly, stacked up in this email.

 

(Oh, by the way, if there is more than one line in the file, even if I used "u" 
for line breaks, it adds a newline to the variable, Is that correct?)

What am I missing? I've opened it in both nano and notepad++, I've used iconv 
and all the tools I can think of to check that file, and all the asterisk conf 
files, too.
It all looks as it should here:


$ cat feature-1.txt
radio$ file feature-1.txt
feature-1.txt: ASCII text
$ wc -l feature-1.txt
0 feature-1.txt
wc -c feature-1.txt
5 feature-1.txt

 

After 6 hours struggling with this, I think I'm starting to lose the plot. Can 
anyone tell me where I'm going wrong? Thanks.



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[asterisk-users] How can I "lock" a device or extension state to only specific states?

[Sebastian Nielsen]

How can I lock a device state so it can only publish AVAILABE, BUSY, or
RINGNING? (Eg, if the device is not BUSY or RINGNING, its AVAILABLE)

 

I have a hint published for a fixed phone and a mobile phone. But if the
mobile phone is out of coverage, off or similar, the queue application will
consider the whole group unavailable.

What I want to check for, is only if the device is BUSY or RINGNING, eg the
device in question is engaged in some sort of call. Then the whole group
should be unavailable when it comes to queues, eg persons in queue has to
wait, because both phones belong to the same person, and the same person
cannot be engaged in 2 calls at once.

 

But if the device is NOT engaged in a call, it should be considered to be
"available", even if the device is offline or not registred, because then
the other device is propably available, and if the "unavailable" device is
that because its offline or not registred, then the person owning it can
obviously not be engaged in the call, and thus its wise to ring the other,
online device.

 

Best regards, Sebastian Nielsen



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[asterisk-users] Queue grouping - how can it be implemented?

[Sebastian Nielsen]

I have a Asterisk set up. In this, I want to use queues.

 

Now I want to group "agents" into groups, such as so if one phone in a group
is busy, the whole group is considered busy.

 

Eg:

Group1:

SIP/Dad

SIP/DadsMobile

 

Group2:

SIP/Mom

SIP/MomsMobile

 

 

If there is three persons in queue, then, then, first, all 4 phones should
ring. Now lets say Mom takes the call via the Mobile.

Now, for the next call in queue, only Dad and DadsMobile should ring. He
picks up the call via the home phone.

 

Now, even if SIP/Mom and SIP/DadsMobile is vacant, both groups should be
considered busy, and the third person in queue, has to wait in queue until
either SIP/MomsMobile or SIP/Dad is complete with the call.

 

How can this be implemented? Can it be implemented with the standard Queue
application through advanced dialplan programming or does it need something
completely custom?



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