Re: [asterisk-users] SIP Source Port

2021-07-12 Thread Sebastian
Maybe it could be accomplished in the firewall? Tell the firewall to NAT the 
source  port of packets to 5061?
 
Från: asterisk-users-boun...@lists.digium.com 
 För Alexander Perkins
Skickat: den 10 juli 2021 19:39
Till: asterisk-users@lists.digium.com
Ämne: [asterisk-users] SIP Source Port
 
Hi All.  We have a provider that requires us to SOURCE the SIP connection on 
TCP 5061.  I honestly have no clue how to force Asterisk to always SOURCE the 
SIP connection on a certain port.  
 
Can anybody point me in the right direction?  I am using PJSIP.
 
Thank you,
Alex


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[asterisk-users] DECT client adapter

2021-03-14 Thread 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

2021-03-11 Thread 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

2021-03-11 Thread 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

2021-03-11 Thread 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

2020-12-31 Thread 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?

2020-10-07 Thread 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?

2020-10-03 Thread 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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[asterisk-users] Inband DTMF not detected - bug or config error?

2020-08-26 Thread Sebastian Damm
Hi,

we have an Asterisk server basically passing on calls using the Dial
application. In the pjsip endpoint settings, the dtmf_mode is set to audio.
This works with most calls. However, there is a scenario where DTMF tones
don't get forwarded the way I would expect them to get forwarded.

A: Caller without RfC4733 support
B: our Asterisk, version 17.6.0
C: Another Asterisk, with RfC4733 support, running an IVR

Now when the call comes in, our Asterisk (B) sends out a new call, offering
telephone-event to C. However, since A and C use the same codec, B bridges
those two calls using a native bridge. And that's probably the reason why
it doesn't detect inband DTMF tones from A, but instead just passes on the
media stream to C. C, however, thinks DTMF should come as RTP events, not
detecting the tones coming inband, either.

The documentation says, the native bridge is not capable of doing anything
with DTMF, but the core bridge would be. (
https://wiki.asterisk.org/wiki/display/AST/Bridges)

In my opinion, Asterisk should not enter native bridge, when dtmf modes for
the two channels differ. Is this a bug? If it isn't, is there a way to
force Asterisk to use the core bridge, so DTMF tones get detected inband
and converted to rtp events?

Any hint appreciated.

Regards,
Sebastian
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Re: [asterisk-users] Channels freeze on Confbridge

2020-08-23 Thread 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

2020-08-22 Thread 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.

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

2020-05-28 Thread 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

2019-11-18 Thread 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
__

Zurlindenstrasse 29 Tel  +41 61 826 93 00
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

2019-11-16 Thread 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

2019-11-16 Thread 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

2019-11-16 Thread 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.

2019-10-14 Thread 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?

2019-03-21 Thread 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?

2019-03-21 Thread 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] Need for more hangup reasons in ARI?

2018-12-06 Thread Sebastian Damm
Hi,

On Thu, Dec 6, 2018 at 2:44 PM Joshua C. Colp  wrote:
> Nope. No specific reason. The ones there are what most people would use, and 
> what those not as familiar with telephony would understand.

Thanks for the quick response. I have submitted a patch for adding
more hangup reasons.

BTW: Is there a way to have them documented on the Wiki page instead
of having to dig into the source code? I'd be happy to help.

Regards,
Sebastian

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[asterisk-users] Need for more hangup reasons in ARI?

2018-12-06 Thread Sebastian Damm
Hi,

while implementing an application based on ARI, I wanted to hangup
calls in different states with different hangup reasons. After not
finding anything in the documentation I found the possible reasons in
the 
sources:(https://github.com/asterisk/asterisk/blob/4ca709768db9bafdfa83ee1cfc6cc0639f097857/res/ari/resource_channels.c#L866)

I think, there are some important reasons missing. For example I want
to return a 408 if the callee doesn't answer after some time. I guess,
I could submit a patch for more hangup reasons, but before doing so,
is there a reason that there are only those five hangup reasons?

Regards,
Sebastian

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

2018-11-15 Thread 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

2018-11-15 Thread 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?

2018-11-12 Thread 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] Writing CDR's to two database servers

2017-06-19 Thread Sebastian Gutierrez
use replication

best regards

> On Jun 19, 2017, at 17:47, Tech Support  wrote:
> 
> All;
> I know that there are probably several solutions to this problem, but 
> what I am trying to do is provide some redundancy for my customers CDR data. 
> I know that doing simple backups of MySQL is probably the easiest way to go, 
> but I’m thinking that there may be some benefit to simultaneously writing the 
> CDR data to multiple servers at once. However, I’m drawing a blank on this 
> one. Has anyone else done this before? Any insight at all would be greatly 
> appreciated.
> Thanks Much;
> John V.
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Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

2017-06-12 Thread Sebastian Gutierrez
same here.

> On Jun 12, 2017, at 10:02, Kseniya Blashchuk  wrote:
> 
> Same about me - need to re-enable membership all the time. Annoying ((
> 
> пн, 12 июн. 2017 г. в 15:59, John Novack  >:
> Not just gmail
> Happening as well with Comcast.net
> 
> My Comcast address is set to forward to another domain, as Comcast seems to 
> now block sending mail with a non Comcast "from" address. they turned that on 
> a couple years ago with no  notice.
> 
> John Novack
> 
> 
> Jonathan H wrote:
>> Me too, also gmail. I emailed the list owner a couple of days ago, but no 
>> reply.
>> 
>> Is everyone else affected also forwarding to another email address
>> (gmail or not)?
>> 
>> Could be wrong, but I'm guessing there may be an incorrect DMARC
>> policy somewhere - although this is the only fail I could find in the
>> headers.
>> 
>> boun...@lists.digium.com ;
>>dmarc=fail (p=NONE sp=NONE dis=NONE) header.from=gmail.com 
>> 
>> 
>> 
>> 
>> On 12 June 2017 at 09:12, Steve Davies  
>>  wrote:
>>> I am also getting this, three or four times in the last month after years of
>>> no problems.
>>> 
>>> I agree that Gmail is the likely common factor, but I would love to have
>>> access to these bounce messages to know whether it is actually an
>>> overly-paranoid list server!
>>> 
>>> Steve
>>> 
>>> On Mon, 12 Jun 2017 at 09:09 Andrew Furey  
>>>  wrote:
 Ditto; a Gmail issue?
 
 Andrew
 
 On 12 June 2017 at 16:00, Marcelo Terres  
  wrote:
> It is happening the same with me.
> 
> Regards,
> Marcelo H. Terres  
> IM: mhter...@jabber.mundoopensource.com.br 
> 
> https://www.mundoopensource.com.br 
> https://twitter.com/mhterres 
> https://linkedin.com/in/marceloterres 
> 
> 
> 
> On 12 June 2017 at 08:07, Olivier  
>  wrote:
>> Hello,
>> 
>> I'm a faithful reader of this mailing list, for several years now.
>> 
>> Lately, I'm receiving emails asking me to re-enable my list
>> subscription due
>> to "excessive bouncing".
>> 
>> What does this exactly mean and why am I receiving this ?
>> Beside re-enabling my subscription, what can I do to improve things ?
>> 
>> Regards
>> 
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Re: [asterisk-users] Callee id over chan_sip trunk

2017-05-15 Thread 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?)

2017-05-11 Thread 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?)

2017-05-10 Thread 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?)

2017-05-10 Thread 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.

2017-04-18 Thread 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] Something similar to Doxygen for standard dialplan?

2017-03-18 Thread Sebastian Gutierrez
Everything is now on release folder on GitHub, documentation and executable.

Hope it helps

On Mar 18, 2017, 20:17 +0100, Sebastian Gutierrez <scg...@gmail.com>, wrote:
> This should work with at least .net framework 4, no dependency needed, just 
> .net framework, I think you should be able to compile it from a vs express 
> version. If you are not able to let me know and next week I will build it for 
> you and upload it as an artifact, in my Astricon 2015 talk (Workflows and 
> Maintainability ) you can also see how to extend this very easily with your 
> custom applications.
>
> Let me know if you need assistance.
>
> Best regards
>
>
> On Mar 18, 2017, 20:13 +0100, Jonathan H <lardconce...@gmail.com>, wrote:
> > Hi, thanks - that looks really good!
> >
> > I was about to embark on some non-visual stuff using Ragic, but this
> > looks great.
> >
> > Is there a binary anywhere, or any instructions to compile? I've never
> > compiled C# code before, and although a quick google suggests it
> > shouldn't be too hard, I might need to know a few things like what
> > version of .net it should be compiled with.
> >
> > The readme just points to the website.
> >
> > Thanks!
> >
> > On 18 March 2017 at 18:57, Sebastian Gutierrez <scg...@gmail.com> wrote:
> > > Check this one:
> > >
> > > https://github.com/IntegraCCS/integradesigner
> > >
> > > You can do many things, document each node, and save xml with each
> > > extension.
> > > We´ve made it open source on Astricon 2015 you can extend it the way you
> > > want.
> > >
> > > Hope it helps you.
> > >
> > > Best regards
> > >
> > >
> > >
> > >
> > > On Mar 18, 2017, 12:50 +0100, Jonathan H <lardconce...@gmail.com>, wrote:
> > >
> > > How are we all documenting complex dialplan?
> > >
> > > Is there something similar to Doxygen?
> > >
> > > I've got around 20 config files covering around 60 contexts and 40
> > > variables. Of course, I've maintained a basic list of the major stuff,
> > > and documented the code throughout, but it's grown to the stage where
> > > it needs to be better documented, have a proper flowchart etc.
> > >
> > > Talking of flowcharts, I see there are several flowchart makers for
> > > Asterisk and other IVRs - specifically, in the flowchart, I need "set
> > > this variable, uses that variable, calls this context, uses that
> > > gosub" and so on.
> > >
> > > So it's not just dragging extensions together.
> > >
> > > Any ideas?!
> > >
> > > Thanks
> > >
> > > --
> > > _
> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > >
> > > Check out the new Asterisk community forum at:
> > > https://community.asterisk.org/
> > >
> > > New to Asterisk? Start here:
> > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> > > --
> > > _
> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > >
> > > Check out the new Asterisk community forum at:
> > > https://community.asterisk.org/
> > >
> > > New to Asterisk? Start here:
> > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > Check out the new Asterisk community forum at: 
> > https://community.asterisk.org/
> >
> > New to Asterisk? Start here:
> > https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Something similar to Doxygen for standard dialplan?

2017-03-18 Thread Sebastian Gutierrez
I´ve just added a .exe to the base folder of the project, check if works for 
you, I will try to attach the pdf where everything is explained, is 25mb I will 
see if allows me to upload it.

Best regards


On Mar 18, 2017, 20:13 +0100, Jonathan H <lardconce...@gmail.com>, wrote:
> Hi, thanks - that looks really good!
>
> I was about to embark on some non-visual stuff using Ragic, but this
> looks great.
>
> Is there a binary anywhere, or any instructions to compile? I've never
> compiled C# code before, and although a quick google suggests it
> shouldn't be too hard, I might need to know a few things like what
> version of .net it should be compiled with.
>
> The readme just points to the website.
>
> Thanks!
>
> On 18 March 2017 at 18:57, Sebastian Gutierrez <scg...@gmail.com> wrote:
> > Check this one:
> >
> > https://github.com/IntegraCCS/integradesigner
> >
> > You can do many things, document each node, and save xml with each
> > extension.
> > We´ve made it open source on Astricon 2015 you can extend it the way you
> > want.
> >
> > Hope it helps you.
> >
> > Best regards
> >
> >
> >
> >
> > On Mar 18, 2017, 12:50 +0100, Jonathan H <lardconce...@gmail.com>, wrote:
> >
> > How are we all documenting complex dialplan?
> >
> > Is there something similar to Doxygen?
> >
> > I've got around 20 config files covering around 60 contexts and 40
> > variables. Of course, I've maintained a basic list of the major stuff,
> > and documented the code throughout, but it's grown to the stage where
> > it needs to be better documented, have a proper flowchart etc.
> >
> > Talking of flowcharts, I see there are several flowchart makers for
> > Asterisk and other IVRs - specifically, in the flowchart, I need "set
> > this variable, uses that variable, calls this context, uses that
> > gosub" and so on.
> >
> > So it's not just dragging extensions together.
> >
> > Any ideas?!
> >
> > Thanks
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > Check out the new Asterisk community forum at:
> > https://community.asterisk.org/
> >
> > New to Asterisk? Start here:
> > https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > Check out the new Asterisk community forum at:
> > https://community.asterisk.org/
> >
> > New to Asterisk? Start here:
> > https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
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Re: [asterisk-users] Something similar to Doxygen for standard dialplan?

2017-03-18 Thread Sebastian Gutierrez
This should work with at least .net framework 4, no dependency needed, just 
.net framework, I think you should be able to compile it from a vs express 
version. If you are not able to let me know and next week I will build it for 
you and upload it as an artifact, in my Astricon 2015 talk (Workflows and 
Maintainability ) you can also see how to extend this very easily with your 
custom applications.

Let me know if you need assistance.

Best regards


On Mar 18, 2017, 20:13 +0100, Jonathan H <lardconce...@gmail.com>, wrote:
> Hi, thanks - that looks really good!
>
> I was about to embark on some non-visual stuff using Ragic, but this
> looks great.
>
> Is there a binary anywhere, or any instructions to compile? I've never
> compiled C# code before, and although a quick google suggests it
> shouldn't be too hard, I might need to know a few things like what
> version of .net it should be compiled with.
>
> The readme just points to the website.
>
> Thanks!
>
> On 18 March 2017 at 18:57, Sebastian Gutierrez <scg...@gmail.com> wrote:
> > Check this one:
> >
> > https://github.com/IntegraCCS/integradesigner
> >
> > You can do many things, document each node, and save xml with each
> > extension.
> > We´ve made it open source on Astricon 2015 you can extend it the way you
> > want.
> >
> > Hope it helps you.
> >
> > Best regards
> >
> >
> >
> >
> > On Mar 18, 2017, 12:50 +0100, Jonathan H <lardconce...@gmail.com>, wrote:
> >
> > How are we all documenting complex dialplan?
> >
> > Is there something similar to Doxygen?
> >
> > I've got around 20 config files covering around 60 contexts and 40
> > variables. Of course, I've maintained a basic list of the major stuff,
> > and documented the code throughout, but it's grown to the stage where
> > it needs to be better documented, have a proper flowchart etc.
> >
> > Talking of flowcharts, I see there are several flowchart makers for
> > Asterisk and other IVRs - specifically, in the flowchart, I need "set
> > this variable, uses that variable, calls this context, uses that
> > gosub" and so on.
> >
> > So it's not just dragging extensions together.
> >
> > Any ideas?!
> >
> > Thanks
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > Check out the new Asterisk community forum at:
> > https://community.asterisk.org/
> >
> > New to Asterisk? Start here:
> > https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > Check out the new Asterisk community forum at:
> > https://community.asterisk.org/
> >
> > New to Asterisk? Start here:
> > https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
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   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Something similar to Doxygen for standard dialplan?

2017-03-18 Thread Sebastian Gutierrez
Check this one:

https://github.com/IntegraCCS/integradesigner

You can do many things, document each node, and save xml with each extension.
We´ve made it open source on Astricon 2015 you can extend it the way you want.

Hope it helps you.

Best regards




On Mar 18, 2017, 12:50 +0100, Jonathan H , wrote:

> How are we all documenting complex dialplan?
>
> Is there something similar to Doxygen?
>
> I've got around 20 config files covering around 60 contexts and 40
> variables. Of course, I've maintained a basic list of the major stuff,
> and documented the code throughout, but it's grown to the stage where
> it needs to be better documented, have a proper flowchart etc.
>
> Talking of flowcharts, I see there are several flowchart makers for
> Asterisk and other IVRs - specifically, in the flowchart, I need "set
> this variable, uses that variable, calls this context, uses that
> gosub" and so on.
>
> So it's not just dragging extensions together.
>
> Any ideas?!
>
> Thanks
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] 14.3.0 download archive corrupt - cannot extract

2017-02-14 Thread Sebastian Gutierrez
is back online now thanks!

On Feb 14, 2017, 11:18 -0300, Joshua Colp <jc...@digium.com>, wrote:
> On Tue, Feb 14, 2017, at 10:13 AM, Sebastian Gutierrez wrote:
> > The 13.14 tar gz doesn’t even exists on the current or in the old
> > releases folder.
> >
> > there seems to be an issue with the latest build not generating the
> > artifacts?
>
> It was temporarily removed during a synchronization but is now back up
> and the issue should be resolved.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] 14.3.0 download archive corrupt - cannot extract

2017-02-14 Thread Sebastian Gutierrez
The 13.14 tar gz doesn’t even exists on the current or in the old releases 
folder.

there seems to be an issue with the latest build not generating the artifacts?

best regards



On Feb 14, 2017, 11:04 -0300, Marcelo Terres , wrote:
> Thanks Joshua.
> Marcelo H. Terres  IM: mhter...@jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
> https://linkedin.com/in/marceloterres
>
>
> On 14 February 2017 at 14:01, Joshua Colp  wrote:
> > On Tue, Feb 14, 2017, at 09:57 AM, Marcelo Terres wrote:
> > > Same problem with me.
> > >
> > > I downloaded the file in 2 different places and had the same error...
> >
> > An issue was filed for tracking this[1] and it will be resolved later
> > today.
> >
> > [1] https://issues.asterisk.org/jira/browse/ASTERISK-26791
> >
> > --
> > Joshua Colp
> > Digium, Inc. | Senior Software Developer
> > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Re: [asterisk-users] semi-OFF-TOPIC - SIP iptables and NAT - same source, different destination

2017-01-27 Thread 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

2016-11-29 Thread 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

2016-11-02 Thread 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

2016-10-20 Thread 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

2016-10-17 Thread 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?

2016-10-15 Thread 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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Re: [asterisk-users] Ast 13.10 to 13.11 stop working webrtc

2016-10-06 Thread Sebastian
the issue is fixed in current trunk head version

El jue., 6 de oct. de 2016 a la(s) 12:07, Sebastian <scg...@gmail.com>
escribió:

> the issue is with chan_sip not on rtp I will check wich commit break this
> and fill an issue.
>
>
> El mié., 5 de oct. de 2016 a la(s) 17:41, Sebastian <scg...@gmail.com>
> escribió:
>
> From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop
> working, failing with
>
> chan_sip.c:4083 retrans_pkt: Hanging up call
> 7238b48c11581d4166b899bf747a05f7@130.211.62.184:0 - no reply to our
> critical packet (see
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
>
>
> is there any way to configure to have the previous behaviour?
> Im trying to set dtlscipher=AES128-SHA but I always see
>
> DTLS ECDH initialized (automatic), faster PFS enabled
>
> any idea?
>
> Thanks!
> res_rtp_asterisk
> --
> * The DTLS part in Asterisk now supports Perfect Forward Secrecy (PFS).
> Enabling PFS is attempted by default, and is dependent on the configuration
> of the module using TLS.
> - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
> specify a ECDHE cipher suite in sip.conf, for example:
> dtlscipher=AES128-SHA
> - Ephemeral DH (DHE) is disabled by default. To enable it, add DH
> parameters
> into the private key file, e.g., sip.conf dtlsprivatekey. For example:
> openssl dhparam -out ./dh.pem 2048
> - Because clients expect the server to prefer PFS, and because OpenSSL
> sorts
>
> its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
> Consider re-ordering your cipher suites in the respective configuration
> file. For example:
> dtlscipher=ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256
> which forces PFS and requires at least DTLS 1.2.
>
>
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Re: [asterisk-users] Ast 13.10 to 13.11 stop working webrtc

2016-10-06 Thread Sebastian
the issue is with chan_sip not on rtp I will check wich commit break this
and fill an issue.


El mié., 5 de oct. de 2016 a la(s) 17:41, Sebastian <scg...@gmail.com>
escribió:

> From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop
> working, failing with
>
> chan_sip.c:4083 retrans_pkt: Hanging up call
> 7238b48c11581d4166b899bf747a05f7@130.211.62.184:0 - no reply to our
> critical packet (see
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
>
>
> is there any way to configure to have the previous behaviour?
> Im trying to set dtlscipher=AES128-SHA but I always see
>
> DTLS ECDH initialized (automatic), faster PFS enabled
>
> any idea?
>
> Thanks!
> res_rtp_asterisk
> --
> * The DTLS part in Asterisk now supports Perfect Forward Secrecy (PFS).
> Enabling PFS is attempted by default, and is dependent on the configuration
> of the module using TLS.
> - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
> specify a ECDHE cipher suite in sip.conf, for example:
> dtlscipher=AES128-SHA
> - Ephemeral DH (DHE) is disabled by default. To enable it, add DH
> parameters
> into the private key file, e.g., sip.conf dtlsprivatekey. For example:
> openssl dhparam -out ./dh.pem 2048
> - Because clients expect the server to prefer PFS, and because OpenSSL
> sorts
>
> its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
> Consider re-ordering your cipher suites in the respective configuration
> file. For example:
> dtlscipher=ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256
> which forces PFS and requires at least DTLS 1.2.
>
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[asterisk-users] Ast 13.10 to 13.11 stop working webrtc

2016-10-05 Thread Sebastian
>From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop
working, failing with

chan_sip.c:4083 retrans_pkt: Hanging up call
7238b48c11581d4166b899bf747a05f7@130.211.62.184:0 - no reply to our
critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).


is there any way to configure to have the previous behaviour?
Im trying to set dtlscipher=AES128-SHA but I always see

DTLS ECDH initialized (automatic), faster PFS enabled

any idea?

Thanks!
res_rtp_asterisk
--
* The DTLS part in Asterisk now supports Perfect Forward Secrecy (PFS).
Enabling PFS is attempted by default, and is dependent on the configuration
of the module using TLS.
- Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
specify a ECDHE cipher suite in sip.conf, for example:
dtlscipher=AES128-SHA
- Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
into the private key file, e.g., sip.conf dtlsprivatekey. For example:
openssl dhparam -out ./dh.pem 2048
- Because clients expect the server to prefer PFS, and because OpenSSL sorts

its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
Consider re-ordering your cipher suites in the respective configuration
file. For example:
dtlscipher=ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256
which forces PFS and requires at least DTLS 1.2.
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[asterisk-users] Queue grouping - how can it be implemented?

2016-06-15 Thread 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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Description: S/MIME Cryptographic Signature
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[asterisk-users] Asterisk registers with TLS, but sends out calls via UDP

2016-05-04 Thread Sebastian Damm
Hi,

I have an Asterisk 13.8.2, which is supposed to be only a client to an
encrypted SIP service. All local phones are connected via UDP.

Since I can't use PJSIP (see my mailing list post from yesterday), I
tried configuring chan_sip to work that way. My settings are:

[general]
context=public
allowoverlap=no
udpbindaddr=0.0.0.
tlsbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
tlsenable=yes
transport=udp
srvlookup=yes
tlscafile=/usr/local/etc/asterisk/keys/4cfd3c78.0
tlscapath=/usr/local/etc/asterisk/keys
tlsclientmethod=tlsv1
sipdebug = yes

register => tls://1234...@example.org:foo...@dev.example.org

[devtrunk]
type=peer
host=example.org
defaultuser=1234567
fromuser=1234567
remotesecret=foobar
transport=tls
outboundproxy=dev.example.org
context=carrier-in
encryption=yes

When I start up, I see my Asterisk doing a _sips._tcp SRV lookup, but
that's just for the registration, I guess. I also see it doing
_sip._udp SRV queries. I wouldn't know why it would have to do that.

The REGISTER packets are sent out via TLS, as I would expect.

When I issue a "sip show peer devtrunk" command, it tells me this:

  Prim.Transp. : TLS
  Allowed.Trsp : TLS

Looks okay to me. But when I place a call, Asterisk does this:

Reliably Transmitting (no NAT) to 2.3.4.5:5060:
INVITE sip:0123456...@example.org SIP/2.0
Via: SIP/2.0/UDP 9.8.7.6:0;branch=z9hG4bK2974d534

It sends the packet out via UDP, and to the wrong host, since it
doesn't use the correct SRV entry and instead sends it to the UDP
server.

I did not generate a certificate for my Asterisk, because it only acts
as a client. I think, this shouldn't be needed.

Can anyone point me to where I misconfigured something? Or did I
stumble upon a bug? What would I have to do to make Asterisk use the
open TLS connection used for registering for outbound calls, too?

Best Regards,
Sebastian

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[asterisk-users] Asterisk (PJSIP) registers with sips Contact URI, but why?

2016-05-03 Thread Sebastian Damm
Hi,

I'm registering an Asterisk against my TLS capable service, using
res_pjsip. My config looks like this:

[devtrunk_reg]
type=registration
outbound_auth=devtrunk_auth
server_uri=sip:example.org\;transport=tls
client_uri=sip:1234...@example.org\;transport=tls
outbound_proxy=sip:dev.example.org\;transport=tls\;lr
contact_user=1234567
retry_interval=60
expiration=600
line=yes
endpoint=222

[devtrunk_auth]
type=auth
auth_type=userpass
username=1234567
password=secret
realm=example.org


It registers fine, but this is what the REGISTER request looks like:

<--- Transmitting SIP request (903 bytes) to TLS:1.2.3.4:5061 --->
REGISTER sip:example.org;transport=tls SIP/2.0
Via: SIP/2.0/TLS
9.8.7.6:55664;rport;branch=z9hG4bKPjNlqlgmSOP7O4LqOTUqJtFZB8fTmc0ZKL;alias
Route: <sip:dev.example.org;transport=tls;lr>
From: <sip:1234...@example.org>;tag=vhDrzKtv9lMR53ZJFgVTnvGcACJiN6Aa
To: <sip:1234...@example.org>
Call-ID: nzgHdLyliuBwecmae2Y..0oY2DqYjH0V
CSeq: 14861 REGISTER
Contact: <sips:1234567@9.8.7.6:55664;transport=TLS;line=dhslasr>
Expires: 600
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, MESSAGE, REFER, REGISTER
Max-Forwards: 70
User-Agent: Asterisk PBX 13.8.2
Content-Length:  0

What I really don't like is the Contact line. It starts with sips
instead of sip. This makes inbound calls not work because the server
sends a sip Contact header instead of sips. And Asterisk rejects that.

In the header of the 480 response I see this line:

Warning: 381 SIP "SIPS Required"

Since I can't reconfigure the server to send sips Contact URIs, I need
Asterisk to send out a contact URI in the register, that starts with
sip: as well. Then inbound calls would work.

Is there any way to get rid of this sips URI?

Interestingly, when sending out calls, the Contact URI starts with sip
instead of sips, so outbound calls work.

Best Regards,
Sebastian

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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

2015-12-22 Thread Sebastian Kemper
On Tue, Dec 22, 2015 at 09:30:52AM +, Luca Bertoncello wrote:
> Zitat von Sebastian Kemper <sebastian...@gmx.net>:
> 
> Hi Sebastian
> 



> 
> I tried with
> 
> sip set debug 42
> sip set verbose 42
> 
> The result was in my first E-Mail...

Hi Luca,

I don't remember seeing anything looking like a SIP trace in your first
mail. Try

sip set debug on

instead of

sip set debug 42

I don't think there's a sip debugging level like 42 in Asterisk. You can
either switch it on or off.

Regards,
Sebastian

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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

2015-12-22 Thread Sebastian Kemper
On Tue, Dec 22, 2015 at 09:42:04AM +, Luca Bertoncello wrote:
> Is it not this:
> 
> http://lists.digium.com/pipermail/asterisk-users/2015-December/288078.html
> 
> ?
> 
> sip set debug 42 should be a little trick to enable more debugging...  
> So I got in this list some months ago...

No, that's not it. SIP debugging should show you all the SIP messages
like INVITEs, ACKs and the likes. See this link:

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Big fat warning: If you want to paste a SIP trace to the mailing list,
make sure to clean it up first (remove passwords, user names, phone
numbers, digest authentication info etc).

Regards,
Sebastian

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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

2015-12-22 Thread Sebastian Kemper
On Tue, Dec 22, 2015 at 07:19:47AM +0100, Luca Bertoncello wrote:
> "Brian ::" <b...@iptel.co> schrieb:
> 
> > sip trace?
> 
> Could you please explain? I'm not a VoIP-expert...
> 
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)

Hi Luca,

Brian suggests to check the SIP traces. You can either enable SIP
debugging in Asterisk like so:

sip set debug on

Or you could run tcpdump and capture the SIP traffic.

The first option is probably the easiest.

Regards,
Sebastian

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[asterisk-users] Re-Invite to Native Bridge

2015-11-06 Thread Sebastian Kemper
Hello all,

My Asterisk is between my ITSP and a SIP phone. I cannot do direct media
between the provider and the SIP phone, but I would like Asterisk to
locally RTP bridge the two channels using native_rtp.

Example:

> Bridge cfb56606-6b40-4da7-a6fa-6499e183cdbb: switching from simple_bridge 
> technology to native_rtp
> Locally RTP bridged 'PJSIP/5iwrlBee9oKCMAs-' and 
> 'PJSIP/ekiga_outbound-0001' in stack
> Locally RTP bridged 'PJSIP/5iwrlBee9oKCMAs-' and 
> 'PJSIP/ekiga_outbound-0001' in stack

My SIP phone supports G722 and PCMA, as does the ITSP provider.  But
depending on the other party in a call, the ITSP may only offer PCMA.

So with my current setting (allow=!all,g722,alaw) I run into this
situation when the ITSP doesn't offer G722:

ITSP <- PCMA -> Asterisk <- G722 -> SIP phone

Obviously that's not optimal. Transcoding needs to take place.

I'm looking for a way to get Asterisk to re-INVITE/UPDATE the SIP phone,
to renegotiate the codec, to avoid transcoding.

I searched the web and found that there are pre-bridge handlers. But
they are executed on the called party channel. So that wouldn't help
when doing an outbound call, plus I'm not sure how I could extract
enough information from the other channel to know if a reinvite is
needed (e.g. the codec the other channel is using).

I also found the Media Format Rewrite article on the wiki:
https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite

In the General Rules paragraph point 3 states (quote):

Prior to entering a bridge, a dialplan function can be used to set
whether or not that channel will attempt to make itself compatible with
whatever is in the bridge with it. If a channel enters a bridge that has
another channel in it with a format it supports, it will attempt to
switch the channel to the bridged channel's format to facilitate native
bridging. Note that this has no bearing in multi-party bridges, where
everyone is transcoded.

There's another paragraph called "Re-Invite to Native Bridge" with an
example where Alice and Bob have a differently ordered set of codecs and
Alice's channel is set to re-INVITE back to native bridging if possible.
After the re-INVITE Asterisk switches to a native bridge.

Well, that is exactly what I want :)

I installed Asterisk 13.6.0 hoping that I could get it to do this. But
until now I haven't found out how.

Does anybody know if this feature from the Media Format Rewrite article
is already available?

Kind regards,
Sebastian

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Re: [asterisk-users] Update peer IP address

2015-09-17 Thread Sebastian Kemper
Am 16. September 2015 18:48:16 MESZ, schrieb Daniel Heckl 
<daniel.he...@gmail.com>:
>Sebastian,
>
>If I have understood you correctly, the SIP communication is now via
>NAT instead forwarded ports. For safety, it is much better.
>
>I think it is not because of a UDP timeout, but rather because of a NAT
>timeout. For this is "qualify" exactly the right thing to let the NAT
>port opened. 
>
>Daniel
Hi Daniel,

Not quite. Asterisk is running on an Openwrt router. So Asterisk is listening 
on a public IP. No NAT involved, no port forwarding.

Openwrt tracks the UDP connection for 180s (default). "qualify" keeps the 
connection alive (every 120s).

Without "qualify" inbound calls wouldn't work starting 180s after the 
registration, until after another 300s, when Asterisk registers again (provider 
requires a registration expiry >480s).

Regards,
Sebastian

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Re: [asterisk-users] Update peer IP address

2015-09-14 Thread Sebastian Kemper
On Tue, Apr 14, 2015 at 08:26:07AM +0200, Sebastian Kemper wrote:
> On Thu, Apr 02, 2015 at 11:33:38PM +0200, Daniel Heckl wrote:
> > I do not want set allowguest=yes. The problem is, there is no official
> > list with ip addresses of Telekom Germany. But I think all ip
> > addresses comes from the ip range 217.0.0.0/13.
> 
> Hello Daniel,
> 
> Judging by the lists I found I think it's more like this subnet:
> 217.0.16.0/255.255.248.0
> 

Hi again,

I had poked a hole in my firewall for packets from above subnet to my
port 5060. I had done the same for the RTP ports.

A while back I realized that for RTP this was nonsense, as the RTP
packets came from servers not even part of the subnet. Although there
were no ports opened for those it just worked, because Telekom is doing
symmetric RTP (and so is Asterisk by default).

So I got rid of the firewall rule that opened the RTP ports. And then it
dawned on me that I don't even need to open the 5060 port. The REGISTER
requests established a UDP connection that the kernel's conntrack module
was tracking anyway. The only issue was that the REGISTERs occurred only
every 480s and the UDP connections were removed after 180s already.

So at first I raised net.netfilter.nf_conntrack_udp_timeout_stream to
500. That worked. But I didn't really want to raise the default. So
instead I added "qualify=yes" to the dtag_inbound peer. Now asterisk is
sending an OPTIONS request to Telekom every 120s (I raised the frequency
from 60 to 120 by setting "qualifyfreq=120" under [general]), which
keeps the connection open.

Just wanted to add that.

Kind regards,
Sebastian

> > I have now the following addition to sip.conf. I think it is the only
> > safe option. Or what would you say?
> > 
> > [telekom](!)
> 
> 
> 
> > [DTAG-IP_IN18_016](telekom)
> > host=217.0.18.16
> > 
> > [DTAG-IP_IN18_036](telekom)
> > host=217.0.18.36
> > 
> > etc.
> 
> This configuration is now running here:
> 
> [general]
>   context=unauthenticated
>   allowguest=no
>   srvlookup=no
>   udpbindaddr=0.0.0.0
>   tcpenable=no
>   localnet=172.16.28.0/24
>   alwaysauthreject=yes
>   directmedia=no
>   sdpsession=MyNewSessionString
>   useragent=MyNewUserAgent
>   language=de
>   tonezone=de
>   defaultexpiry=480
> 
> register => 0NUMBER2:PASS:u...@t-online.de@tel.t-online.de/NUMBER2
> register => 0NUMBER3:PASS:u...@t-online.de@tel.t-online.de/NUMBER3
> register => 0NUMBER4:PASS:u...@t-online.de@tel.t-online.de/NUMBER4
> 
> [my-codecs](!)
>   allow=!all,alaw
> 
> [home-phone](!,my-codecs)
>   acl=voice_vlan
>   type=friend
>   host=dynamic
>   context=LocalSets
> 
> [XXX](home-phone)
>   secret=X
> 
> [dtag_inbound](my-codecs)
>   acl=acl_dtag_inbound
>   type=peer
>   context=from_dtag
>   host=tel.t-online.de
> 
> [dtag_outbound](my-codecs)
>   acl=acl_dtag_outbound
>   type=peer
>   defaultuser=u...@t-online.de
>   remotesecret=PASS
>   host=tel.t-online.de
>   fromdomain=tel.t-online.de
> 
> The thing is, the provider's SIP server hasn't changed the IP yet. This
> morning it's still the same as yesterday. And yesterday it was the same
> the whole day.
> 
> Don't know why I didn't run into the "load balancer" issue (yet). I'm
> starting to think it's because I'm "hiding" that I'm using Asterisk
> (sdpsession, useragent, also custom systemname in asterisk.conf). But
> probably that's not the reason. Anyway, I'm just going to wait until it
> doesn't work and then worry about it.
> 
> Regards,
> Sebastian

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Re: [asterisk-users] Asterisk 11 and pulseaudio setup as local user

2015-07-03 Thread Sebastian Kemper
Am 3. Juli 2015 13:17:34 MESZ, schrieb Jerry Geis ge...@pagestation.com:
alsa_card_init^[[0m: snd_pcm_open failed: Connection refused
soundcard_init^[[0m: Problem opening alsa capture device

These are the errors I get.

I changed the following:
chown -R myuser:myuser  /var/log/asterisk
chown -R myuser:myuser /var/lib/asterisk
chown -R myuser:myuser /var/spool/asterisk
chown -R myuser:myuser /var/run/asterisk
chown -R myuser:myuser /usr/lib/asterisk
edited asterisk.conf and uncommented runuser and rungroup and changed
to
myuuser

Then I get the above error when running


Jerry




Sounds more like wrong perms on the sound devices in /dev. Is your user in the 
audio group, or whatever it's called on your box?

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Re: [asterisk-users] Logging in local time

2015-06-05 Thread Sebastian Kemper
Am 5. Juni 2015 16:29:21 MESZ, schrieb Luca Bertoncello lucab...@lucabert.de:
Hi again!

I just noticed, that my Asterisk (running on an OpenWRT-Switch) writes 

the logs using GMT...
On the Switch the time is right configured and a date says me the  
current LOCAL time.

I didn't found in logger.conf or other file an option to set the
timezone.
Can someone help me?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

Hi Luca,

set up a proper /etc/timezone, see 
http://wiki.openwrt.org/doc/howto/voip.asterisk.

Regards,
Sebastian

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Re: [asterisk-users] Signaling incoming call

2015-05-31 Thread Sebastian Kemper
Am 31. Mai 2015 10:58:56 MESZ, schrieb Luca Bertoncello lucab...@lucabert.de:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Guenther Boelter gboel...@gmail.com schrieb:

 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA256
 
 On 05/31/2015 02:31 PM, Luca Bertoncello wrote:
  Hi list!
  
  Now all works as expected, at least in the simulation I did with
  AsteriskNOW. Hopefully it will work later, when Deutsche Telekom
  changes my ISDN to VoIP...
 
 Don't worry, Asterisk works very well with Deutsche Telekom and there
 new ip-based connections ...

That's a good news...
Currenty I configured my sip.conf so:

register = 004935:MYSECRET@pbxluca/004935
register = 0049351222:MYSECRET@pbxfax/0049351222
register = 0049351333:MYSECRET@pbxanika/0049351333
register = 44:MYVERYSECRET@messagenet/44

[pbxluca]
type=peer
defaultuser=004935
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=luca_incoming 
outboundproxy=172.16.34.132
port=5060
fromuser=004935
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=600

[pbxfax]
type=peer
defaultuser=0049351222
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=fax_incoming 
outboundproxy=172.16.34.132
port=5060
fromuser=0049351222
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=600

[pbxanika]
type=peer
defaultuser=0049351333
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=anika_incoming 
outboundproxy=172.16.34.132
port=5060
fromuser=0049351333
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=600

[messagenet]
type=peer
defaultuser=44
secret=MYVERYSECRET
dtmfmode=rfc2833
host=sip.messagenet.it
context=messagenet_incoming 
outboundproxy=sip.messagenet.it
port=5061
fromuser=44
fromdomain=sip.messagenet.it
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=60

Am I right if I say, that I just have to change defaultuser, host,
secret, outboundproxy and fromdomain with the data from Telekom
and it
works?

I thinks, it should be:

defaultuser=0049351333
secret= MYSECRET
host=tel.t-online.de
context=anika_incoming 
outboundproxy=tel.t-online.de
port=5060
fromuser=0049351333
fromdomain=tel.t-online.de

I'm not sure, where I should write my Login (from my DSL-Line)...
I see this page (in German):

http://hilfe.telekom.de/hsp/cms/content/HSP/de/3378/FAQ/theme-133631783/Auftrag/theme-82239611/IP-basierter-Anschluss/faq-350884716;jsessionid=A18F587E00F25C8FC26ACF3685481D72

Could you please help me?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.10 (GNU/Linux)

iEYEARECAAYFAlVqzVAACgkQ8Ggznj+1EDifYwCgiQTeZQsUljAP5CNpteeFW5aV
ugMAn0BnmlGJRHqBJA19DXPgqv0ZUqq1
=vt3E
-END PGP SIGNATURE-

Hi Luca,

I had a discussion recently regarding Asterisk and your provider. The result 
you can basically find in this message: 
http://lists.digium.com/pipermail/asterisk-users/2015-April/286353.html

Regards,
Sebastian

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Re: [asterisk-users] Debugging dialplan

2015-05-29 Thread Sebastian Kemper
On Fri, May 29, 2015 at 07:24:45AM +0200, Luca Bertoncello wrote:
 
 Since I think, I have a problem in my dialplan, how can I debug it?
 It would be very useful a command in Asterisk CLI to ask Asterisk what it
 would do if the number X call the number Y.
 Something like exim -bt, if someone here know the SMTP-daemon Exim...
 
 Is there such an option in Asterisk?
 

Hi Luca,

try 'dialplan show number@context'.

Regards,
Sebastian

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Re: [asterisk-users] Debugging dialplan

2015-05-29 Thread Sebastian Kemper
Hi Luca,

It's not the A number you have to look at if you want to know how a call comes 
into the dialplan and then goes out again. You want do know in which context a 
call arrives. That depends on things like the IP address (peer), 
username/password (friend) or other things.

I suggest to read up on that using the Internet (there are e.g. wiki articles 
about this subject) or a book (e.g. Definitive Guide on Asterisk).

Regards,
Sebastian

Am 29. Mai 2015 09:42:55 MESZ, schrieb Luca Bertoncello lucab...@lucabert.de:
Zitat von jg webaccounts...@jgoettgens.de:

 Yes, it is called core set verbose 42, the other options is core  
 set debug 42.  Enjoy the show!

OK, thanks, but with this option I can just debug what happens if I  
call an extension right now...
I'd like to have a command to ask Asterisk how it will handle a call...

 Once you are more familiar with *, you might want to have a look  
 what you can do with logger.conf.

Maybe later...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)


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Re: [asterisk-users] Update peer IP address

2015-04-14 Thread Sebastian Kemper
On Thu, Apr 02, 2015 at 11:33:38PM +0200, Daniel Heckl wrote:
 I do not want set allowguest=yes. The problem is, there is no official
 list with ip addresses of Telekom Germany. But I think all ip
 addresses comes from the ip range 217.0.0.0/13.

Hello Daniel,

Judging by the lists I found I think it's more like this subnet:
217.0.16.0/255.255.248.0

 I have now the following addition to sip.conf. I think it is the only
 safe option. Or what would you say?
 
 [telekom](!)

snip

 [DTAG-IP_IN18_016](telekom)
 host=217.0.18.16
 
 [DTAG-IP_IN18_036](telekom)
 host=217.0.18.36
 
 etc.

This configuration is now running here:

[general]
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=0.0.0.0
tcpenable=no
localnet=172.16.28.0/24
alwaysauthreject=yes
directmedia=no
sdpsession=MyNewSessionString
useragent=MyNewUserAgent
language=de
tonezone=de
defaultexpiry=480

register = 0NUMBER2:PASS:u...@t-online.de@tel.t-online.de/NUMBER2
register = 0NUMBER3:PASS:u...@t-online.de@tel.t-online.de/NUMBER3
register = 0NUMBER4:PASS:u...@t-online.de@tel.t-online.de/NUMBER4

[my-codecs](!)
allow=!all,alaw

[home-phone](!,my-codecs)
acl=voice_vlan
type=friend
host=dynamic
context=LocalSets

[XXX](home-phone)
secret=X

[dtag_inbound](my-codecs)
acl=acl_dtag_inbound
type=peer
context=from_dtag
host=tel.t-online.de

[dtag_outbound](my-codecs)
acl=acl_dtag_outbound
type=peer
defaultuser=u...@t-online.de
remotesecret=PASS
host=tel.t-online.de
fromdomain=tel.t-online.de

The thing is, the provider's SIP server hasn't changed the IP yet. This
morning it's still the same as yesterday. And yesterday it was the same
the whole day.

Don't know why I didn't run into the load balancer issue (yet). I'm
starting to think it's because I'm hiding that I'm using Asterisk
(sdpsession, useragent, also custom systemname in asterisk.conf). But
probably that's not the reason. Anyway, I'm just going to wait until it
doesn't work and then worry about it.

Regards,
Sebastian

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Re: [asterisk-users] Update peer IP address

2015-04-14 Thread Sebastian Kemper
On Tue, Apr 14, 2015 at 09:38:22AM +0200, Daniel Heckl wrote:
 Sebastian,
 
 Your code sounds good, I'm curious how it goes on.
 
 First the linux machine had the Google Public DNS 8.8.8.8 as DNS
 server. After I changed it to the via PPPoE assigned DNS servers, i
 had no changes any more. But we should be prepared for changes.
 
 You must enable the dnsmgr. If DNS resolves a new ip, the peer is
 updated.

Hello Daniel,

Thanks for the tip. I've enabled the DNS manager. Let's see how it goes.

Kind regards,
Sebastian

 
  Am 14.04.2015 um 08:26 schrieb Sebastian Kemper sebastian...@gmx.net:
  
  On Thu, Apr 02, 2015 at 11:33:38PM +0200, Daniel Heckl wrote:
  I do not want set allowguest=yes. The problem is, there is no official
  list with ip addresses of Telekom Germany. But I think all ip
  addresses comes from the ip range 217.0.0.0/13.
  
  Hello Daniel,
  
  Judging by the lists I found I think it's more like this subnet:
  217.0.16.0/255.255.248.0
  
  I have now the following addition to sip.conf. I think it is the only
  safe option. Or what would you say?
  
  [telekom](!)
  
  snip
  
  [DTAG-IP_IN18_016](telekom)
  host=217.0.18.16
  
  [DTAG-IP_IN18_036](telekom)
  host=217.0.18.36
  
  etc.
  
  This configuration is now running here:
  
  [general]
 context=unauthenticated
 allowguest=no
 srvlookup=no
 udpbindaddr=0.0.0.0
 tcpenable=no
 localnet=172.16.28.0/24
 alwaysauthreject=yes
 directmedia=no
 sdpsession=MyNewSessionString
 useragent=MyNewUserAgent
 language=de
 tonezone=de
 defaultexpiry=480
  
  register = 0NUMBER2:PASS:u...@t-online.de@tel.t-online.de/NUMBER2
  register = 0NUMBER3:PASS:u...@t-online.de@tel.t-online.de/NUMBER3
  register = 0NUMBER4:PASS:u...@t-online.de@tel.t-online.de/NUMBER4
  
  [my-codecs](!)
 allow=!all,alaw
  
  [home-phone](!,my-codecs)
 acl=voice_vlan
 type=friend
 host=dynamic
 context=LocalSets
  
  [XXX](home-phone)
 secret=X
  
  [dtag_inbound](my-codecs)
 acl=acl_dtag_inbound
 type=peer
 context=from_dtag
 host=tel.t-online.de
  
  [dtag_outbound](my-codecs)
 acl=acl_dtag_outbound
 type=peer
 defaultuser=u...@t-online.de
 remotesecret=PASS
 host=tel.t-online.de
 fromdomain=tel.t-online.de
  
  The thing is, the provider's SIP server hasn't changed the IP yet. This
  morning it's still the same as yesterday. And yesterday it was the same
  the whole day.
  
  Don't know why I didn't run into the load balancer issue (yet). I'm
  starting to think it's because I'm hiding that I'm using Asterisk
  (sdpsession, useragent, also custom systemname in asterisk.conf). But
  probably that's not the reason. Anyway, I'm just going to wait until it
  doesn't work and then worry about it.
  
  Regards,
  Sebastian

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Re: [asterisk-users] Update peer IP address

2015-04-01 Thread Sebastian Kemper
On Tue, Mar 31, 2015 at 12:36:34PM +0200, Daniel Heckl wrote:
 Hello Sebastian,
 
 I had already seen this list of the hosts, but it is not active. All
 servers with which my Asterisk has been communicated are not listed.
 
 A port scan, to eventually update the list, found hundreds of servers
 provided in the address range 217.0.0.0/13 with open port 5060, some
 were even not found. I think there must be another solution.
 
 If I change insecure to insecure=port,invite - could that be a
 solution?

Hello Daniel,

I've asked myself that, too. But I don't have access to the connection,
yet, so I can't test it right away.

Kind regards,
Sebastian

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Re: [asterisk-users] Update peer IP address

2015-04-01 Thread Sebastian Kemper
On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote:
 On 4/1/15 10:48 AM, Daniel Heckl wrote:
  John,
 
  thank you four your answer. I think you have misunderstood the
  problem. It’s about a ip address change of the sip trunk, not of my
  asterisk server.
 You would probably benefit by enabling the DNS Manager to allow for
 dynamic IP changes:
 
 # cat dnsmgr.conf [general] enable=yes ; enable creation
 of managed DNS lookups ;   default is 'no' refreshinterval=180   ;
 refresh managed DNS lookups every n seconds ;   default is 300 (5
 minutes)

Hello Andres,

I read that same suggestion elsewhere in connection with Deutsche
Telekom, so it seems there's some benefit in it.

Daniel, did you try it out already?

Kind regards,
Sebastian

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Re: [asterisk-users] Update peer IP address

2015-03-30 Thread Sebastian Kemper
On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote:
 Hello
 
 I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom
 Germany. We have sometimes problems with incoming and outgoing calls.
 I hope I can explain it understandable.

Hello Daniel,

I'll find myself in the same situation a few weeks from now :-)

 
 For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de
 http://tel.t-online.de/), the message is answered with OK and the
 peer is registered.
 
 Usually INVITES comes now from this ip address. All works fine. But
 sometimes INVITES comes from an other IP address, for example
 217.0.23.100. This request Asterisk responds with 401 Unauthorized.
 
 In the next register procedure REGISTER are sent to the new ip address
 and answered also with OK. But qualify OPTIONS are continue be sent to
 the old ip address. Incoming and outgoing calls are canceled. Outgoing
 calls are answered with Forbidden.
 
 Even if the REGISTER procedure works with the new ip address, the
 peers are connected with the old address.
 
 Waiting doesn’t help, only a „sip reload“ update the ip address of the
 peer. 
 
 What is the solution for this problem? How can asterisk update the
 peer?

I think the solution - for the inbound issue at least - could be to add
more hosts as a peer. Have a looks at this forum post:

http://www.ip-phone-forum.de/showthread.php?t=268787p=1999371viewfull=1#post1999371

The user used a template and than he added peers, each with its own IP
address. The provided list was last updated in 2014, though, so I assume
the provider in the meantime has added to that list.

It looks pretty tedious, though, I mean there could be dozens of IPs
you'd have to add. But I guess this is the way to go with Asterisk 11
and chan_sip.

The future looks brighter :-) I read that with pjsip, which I understand
is the replacement for chan_sip, you can have one peer entry and match
an IP range instead of a single host. That should tidy up the dialplan.

What I'm a little afraid of is the SIP provider using IPs out of a range
that they also use for other services. Maybe out of the same range they
hand out IPs to their customers. I guess we got to be careful :-)

Kind regards,
Sebastian

 The Asterisk is local behind a NAT with a firewall, following settings
 are used:
 
 externhost with DynDNS stun with stun.t-online.de
 http://stun.t-online.de/ nat=yes srvlookup=yes allowguest=no
 trustrpid=no insecure=invite qualify=yes
 
 Thank you!  Daniel

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[asterisk-users] Asterisk on OpenWrt (first time user)

2015-03-20 Thread Sebastian Kemper
Hello list,

I'm hoping that you could read through this mail and give me some tips
on how to improve my setup (functionality, security, really anything).
It's my first Asterisk installation and meant for simple home use.

I installed Asterisk 11 on an OpenWrt Barrier Breaker router. Currently
it's configured for Ekiga so I can test. In a few weeks I'll change to a
Telco SIP provider for a PSTN connect.

My Ekiga test calls are successful. So it does seem to work :)

The router is configured like this:

- has a user 'asteriskpbx' so Asterisk doesn't run as root
- has a USB stick for logs and CDRs
- has a LAN port which I removed from the regular LAN VLAN and put into
  its own VLAN; only the hardware SIP phone is connected; there's no
  forwarding to and from this network
- firewall allows SIP and RTP packets from the outside, but only from
  Ekiga.net

The main config file:

[directories]
astdbdir = /mnt/usb/asterisk/dbdir
astlogdir = /mnt/usb/asterisk/logdir

[options]
systemname = my.dynamic.domain.com  
runuser = asteriskpbx   
rungroup = asteriskpbx  
defaultlanguage = de   
documentation_language = en_US  
live_dangerously = no   

[compat]
pbx_realtime=1.6
res_agi=1.6
app_set=1.6

Here's the SIP setup:

[general]
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=0.0.0.0
tcpenable=no
localnet=172.16.28.0/24
alwaysauthreject=yes
language=de

register = MyEkigaUser:myekigap...@ekiga.net/MyEkigaUser

[my-codecs](!)
allow=!all,alaw

[home-phone](!,my-codecs)
acl=voice_vlan
directmedia=no
type=friend
host=dynamic
context=LocalSets

[MyPhoneMacAddress](home-phone)
secret=MyPhonePassword

[ekiga_inbound](my-codecs)
acl=acl_ekiga_inbound
type=peer
host=ekiga.net
context=from-ekiga

[ekiga_outbound](my-codecs)
acl=acl_ekiga_outbound
type=peer
host=ekiga.net
defaultuser=MyEkigaUser
remotesecret=MyEkigaPass
fromuser=MyEkigaUser
fromdomain=ekiga.net

This is my dialplan:

[LocalSets]
exten = 101,1,Dial(SIP/MyPhoneMacAddress,30)
exten = 500,1,Dial(SIP/ekiga_outbound/500,30)
exten = 501,1,Set(GROUP(users)=CallsToProvider)
 same = n,NoOp(There are ${GROUP_COUNT(CallsToProvider)} calls for 
account CallsToProvider.)
 same = n,GotoIf($[${GROUP_COUNT(CallsToProvider)}  
1]?denied:continue)
 same = n(denied),NoOp(There are too many calls up already. Hang up.)
 same = n,HangUp()
 same = n(continue),NoOp(Continue processing call as normal here ...)
 same = n,Dial(SIP/ekiga_outbound/501,30)
exten = 520,1,Dial(SIP/ekiga_outbound/520,30)

; For later when dialing out via the PSTN connect
;
; Apparently the German RegTP says that local numbers are at least
; 4 digits long. So the following will catch all calls (1234, 0242113,
; 003412314 etc.). Lets strip everything except numbers from the
; extension, though. And only allow 1 call at a time.
;
;   exten = _.,1,Set(GROUP(users)=CallsToProvider)
;same = n,NoOp(There are ${GROUP_COUNT(CallsToProvider)} calls from 
LocalSets to sip_provider_out.)
;same = n,GotoIf($[${GROUP_COUNT(CallsToProvider)}  
1]?denied:continue)
;same = n(denied),NoOp(There are too many calls up already. Hang up.)
;same = n,HangUp()
;same = n(continue),NoOp(Continue processing call as normal here ...)
;same = n,Set(SAFE_EXTEN=${FILTER(0-9,${EXTEN})})
;same = n,Set(CALLERID(name)=area code + number)
;same = n,Set(CALLERID(num)=area code + number)
;same = n,Dial(SIP/sip_provider_out/${SAFE_EXTEN},30)
;
; Emergeny numbers (110, 112 and 115):
;
;   exten = _11[025],1,Set(CALLERID(name)=area code + number)
;same = n,Set(CALLERID(num)=area code + number)
;same = n,Dial(SIP/sip_provider_out/${EXTEN})

[from-ekiga]
exten = MyEkigaUser,1,Dial(SIP/MyPhoneMacAddress,30)

The AMI Manager is disabled, as is CLI access for anybody except root:

[general]
default_perm = deny
[root]
permit = all

Named ACLs:

[voice_vlan]
deny=0.0.0.0/0.0.0.0
permit=172.16.28.0/255.255.255.0

[acl_ekiga_inbound]
deny=0.0.0.0/0.0.0.0
permit=86.64.162.35

[acl_ekiga_outbound]
deny=0.0.0.0/0.0.0.0

Only load necessary modules:

[modules]
autoload=no
load = chan_sip.so
load = res_rtp_asterisk.so
load = app_dial.so
load = pbx_config.so
load = app_cdr.so
load = cdr_csv.so
load = func_strings
load = func_groupcount.so

Any tips/hints/suggestions appreciated. Thanks for reading!

Kind regards,
Sebastian

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[asterisk-users] [PoE] Avaya 1152a1x

2015-03-16 Thread Sebastian Niehaus
Sorry for posting slightly off topic,

I use an Avaya 1152a1x midspan device to deliver power some Snom phones.
The Power distrubution unit seems to be identical to  PowerDsine
PD-6024G/AC/M.

I can get some basic information via SNTP (networking setup, uptime) but
no port monitoring and no monitoring of power output using snmpwalk on
the device.


Does anyone has experience wit such an device? Does anyone know what my
fault might be? How I can configure the device to output more
information via SNMP?


Thank you very much!


Sebastian



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Re: [asterisk-users] constantly increasing load in Asterisk 11.14

2015-02-06 Thread Sebastian Damm
Hi,

On Thu, Feb 5, 2015 at 4:56 PM, Scott Griepentrog sgriepent...@digium.com
wrote:

 Can you tell me if the memory usage by Asterisk is also increasing with
 load over time?


Yes, the memory usage does rise a bit.

USER   PID %CPU %MEMVSZ   RSS TTY  STAT START   TIME COMMAND
11.14
root 26047  8.9  5.0 1302036 407640 ?  Sl   Jan28 1201:26 asterisk
-vvvgf
11.6
root 36336  9.9  1.1 881684 90360 ?Sl   Jan28 1328:19 asterisk
-vvvgf

Interestingly, the CPU graphs show no visible increase.

If you need any more information, just let me know.

Gareth, the slow increase of load over months was there with 11.5 or 11.6
already, but I can live with a restart every couple month. Once every week
is too much.
Here is another overview of the load of one machine: http://pbrd.co/1zeFwBy
You can see, the Asterisk was running from April to September without much
change in load, then was restarted. In November, we updated to 11.14, and
from that time, it looks a bit different (and Asterisk needed a lot more
restarts).

Best Regards,
Sebastian
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[asterisk-users] constantly increasing load in Asterisk 11.14

2015-02-05 Thread Sebastian Damm
Hi,

we have quite a few Asterisk machines running and try to keep them on a
current version of the Asterisk 11 branch. But since we upgraded to 11.14.0
a couple weeks ago, we have to restart the Asterisk process every week
because the load gets too high and our monitoring complains.

Those machines are doing only SIP-to-SIP call relay, the dialplan is quite
complex, transcoding is done only on a few percent of the calls processed.
During the daytime, there are at max around 200 SIP channels (100 calls)
running at the same time. After one week, one machine has processed about
170k calls.

I have uploaded a comparison of cacti load graphs for one week of a machine
running with 11.14.0 and one running with 11.6.0: http://pbrd.co/1v0SO3R

As you can see, after a restart, both machines have about the same load.
But after the really quiet weekend, the 11.14 Asterisk starts the new week
with a much higer load than the 11.6 Asterisk, where it stays constant.
We've had an 11.5.1 machine running for about half a year without the need
of restarting, but right now, this is not possible.

Has anyone seen this before? Or does anyone know a reason, what change
somewhere between 11.6 and 11.14 could cause this behaviour? It looks like
we have to go back to 11.6.

Best Regards,
Sebastian
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Re: [asterisk-users] Multicast AMI?

2014-10-11 Thread Sebastian
have you seen astmanproxy?

best regards

On Sep 23, 2014, at 10:05, jg webaccounts...@jgoettgens.de wrote:

 Hi!
 
 Maybe I have overlooked something, but I am sort of facing the following 
 problem. I always used the AMI interface to allow (older) client programs on 
 Windows to use their TAPI client code in order to communicate with Asterisk 
 servers. The functionality is basically minimal as only incoming calls need 
 to get detected and there are occasional outgoing ones.
 
 For the outgoing calls, a regular AMI connection is fine, but having a lot of 
 machines ( 20) maintaining separate TCP connection, just to listen to some 
 events seems to be too much effort. Does a (read-only) multicast for relevant 
 events make any sense (except for data protection issues)?
 
 I am considering writing a small proxy that does exactly that with 
 correspondig changes for my Telephony Service Provider (TSP). Would it make 
 any sense to implement this as an option for the Asterisk program? Basically, 
 Actions and Action responses via TCP and Events via multicast?
 
 jg
 
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Re: [asterisk-users] Asterisk and hylafax: how to debug ...

2013-05-23 Thread Sebastian Niehaus
Am 08.05.2013 01:12, schrieb James Cloos:
 SN == Sebastian Niehaus nieh...@web.de writes:
 
 SN Running Asterisk (version: 1.8.13.1~dfsg-3) on Debian Wheezy. On the
 SN same maschine: Hylafax fax server. I want hylafax to use t38modem (a
 SN virtual T.38 modem) for sending faxes. t38modem schould connect to
 SN asterisk on the same host.
 
 SN If hylafax sends a fax it should use t38modem which ist connected to
 SN asterisk. Asterik is expected to establish an outbound connection to my
 SN SIP provider which supports T38. The asterisk box is behind nat.
 
 Silly question:  

Not so silly ...

 If you want to use T38 to the remote provider, and have
 t38modem, do you /need/ the asterisk in the middle?

The hylafax server is behind NAT and I did not succeed to get t38modem
running behind NAT. So I wanted to give asterisk a try since it has some
features to make it work behind a natted connection.


 And if you /do/ need something between the two, might a sip proxy work
 better than a pbx

I am behind a IAD which does NAT. I cannot install my own software on
the IAD. Therefore I did nit consider a SIP proxy.


Thanks!


Sebastian



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[asterisk-users] Asterisk and hylafax: how to debug ...

2013-05-07 Thread Sebastian Niehaus
Hi,

I hope you might give me some hints on how to find where my
configuration is wrong, I am new to Asterisk and do not know, how to
find the problem.


Running Asterisk (version: 1.8.13.1~dfsg-3) on Debian Wheezy. On the
same maschine: Hylafax fax server. I want hylafax to use t38modem (a
virtual T.38 modem) for sending faxes. t38modem schould connect to
asterisk on the same host.

If hylafax sends a fax it should use t38modem which ist connected to
asterisk. Asterik is expected to establish an outbound connection to my
SIP provider which supports T38. The asterisk box is behind nat.

For some reason, t38modem tells hylafax the line is BUSY so there is no
fax send.
I don't know why there is a busy signal, maybe the call forwarding
configuration is wrong, maybe the registration on my SIP provider fails,
maybe ?

I simply don't know how to debug what's going on. If Asterix trying to
establish an outgoing connection ... Maybe you can help to enlighten me :-)



-[ sip.conf ]--


[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
localnet=192.168.0.0/255.255.255.0
localnet=127.0.0.1
externhost=hostname.no-ip.org
;disallow=all
;allow=ulaw
;allow=alaw
language=de
nat=yes


; incoming
register = 495361000:passw...@sip.1und1.de/495361000






; local SIP-Account where t30modem registers
[30]
callerid=T38modem30
host=dynamic
;domain=127.0.0.1
;host=  127.0.0.1
:permit=127.0.0.1
user=30
secret=password
type=friend
;mailbox=30
nat=no
context=fax_out
;port=6060
canreinvite=no
t38pt_udptl=yes





[20]
; FritzBox
; this is an ATA, but this entry is
; probably not needed; the ATA does not register
; a SIP account on asterisk.
callerid=FritzBox20
type=friend
username=20
secret=password
host=192.168.0.222
fromuser=20
canreinvite=no
qualify=no
disallow=all
allow=alaw
allow=ulaw
;allow=ilbc
allow=g726
;allow=g729
allow=gsm
;insecure=very
nat=no
dtmfmode=info
;tos=0x18






; Outgoing calls to my SIP provider
[495361000]

type=friend
username=495361000
secret=password
host=sip.1und1.de
fromuser=495361000
canreinvite=no
qualify=no
disallow=all
allow=alaw
allow=ulaw
;allow=ilbc
allow=g726
;allow=g729
allow=gsm
;insecure=very
nat=yes
dtmfmode=info
tos=0x18


-[ end of sip.conf ]---


-[ extensions.conf ]---


[general]
static=yes
writeprotect=no




[1und1-fax-out]
exten = _0.,1,Dial,SIP/${EXTEN}@495361000|45|r


[default]
include = 1und1-fax-out

-[ end of extensions.conf ]




Any idea what might be wrong?


Thank you very much!


Sebastian


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Re: [asterisk-users] Asterisk and hylafax: how to debug ...

2013-05-07 Thread Sebastian Niehaus
Am 07.05.2013 18:23, schrieb Sebastian Niehaus:

 For some reason, t38modem tells hylafax the line is BUSY so there is no
 fax send.

Well, I may add the log of t38modem (sorry for the ugly formatting)
Parts I consider as most important are:


 ModemConnection::SetUpConnection dstNum=189659 srcNum=30 srcName=root

... denied (all modems busy)

SIP OnSetUp failed for INVITE from sip:06814003340@127.0.0.1:6060
for Call[Cf64671db22]-EPsip
ModemConnection::OnReleased Call -EPmodem[modem:/C8670792521/0] (Call
cleared because the line is out of service)



Thanks for any suggestions 


here is all the log:
---

2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700 -- AT+FCLASS=1
2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700 --  {
  0d 0a 4f 4b 0d 0a  ..OK.. }
2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700 -- ATDT0681/4003340
2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700
ModemEndPoint::OnMyCallback command=dial extra=5
2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700 PseudoModemQ::Dequeue
ttyT38-0
2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700 MyManager::OnMyCallback
SetUpCall(modem:, 06814003340@+/dev/ttyT38-0)
2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700
ModemEndPoint::MakeConnection modem:
2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700
ModemConnection::SetUpConnection
Call[Cdd70c36e1]-EPmodem[modem:/Cdd70c36e1/0]
Call[Cdd70c36e1] from modem:T38modem to 06814003340@+/dev/ttyT38-0,
route to sip:06814003340@127.0.0.1:6060
2013/05/07 21:30:03.575 ttyT38-0(e...0xb1778700 Call[Cdd70c36e1] from
modem:T38modem to 06814003340@+/dev/ttyT38-0, route to
sip:06814003340@127.0.0.1:6060
2013/05/07 21:30:03.576 ttyT38-0(e...0xb1778700
ModemEndPoint::GetMediaFormats
2013/05/07 21:30:03.585 ttyT38-0(e...0xb1778700
ModemEndPoint::OnMyCallback request={
calltoken=modem:/Cdd70c36e1/0
localpartyname=
command=dial
response=confirm
number=06814003340
modemtoken=ttyT38-0
}
2013/05/07 21:30:03.588 Pool:0xb1674700
ModemEndPoint::MakeConnection modem:06814003340
Call[C8e34455a2] from sip:06814003340@127.0.0.1:6060 to
sip:06814003340@127.0.0.1:6060, route to modem:06814003340
2013/05/07 21:30:03.588 Pool:0xb1674700 Call[C8e34455a2] from
sip:06814003340@127.0.0.1:6060 to sip:06814003340@127.0.0.1:6060, route
to modem:06814003340
2013/05/07 21:30:03.588 Pool:0xb1674700
ModemEndPoint::GetMediaFormats
2013/05/07 21:30:03.593 Pool:0xb1674700
ModemConnection::SetUpConnection
Call[C8e34455a2]-EPmodem[modem:06814003340/C8e34455a2/0]
2013/05/07 21:30:03.593 Pool:0xb1674700
ModemConnection::SetUpConnection dstNum=06814003340 srcNum=30
srcName=root ...
2013/05/07 21:30:03.593 Pool:0xb1674700 ... denied (all modems busy)
2013/05/07 21:30:03.593 Pool:0xb1674700
ModemConnection::OnReleased
Call[C8e34455a2]-EPmodem[modem:06814003340/C8e34455a2/0]
2013/05/07 21:30:03.593 Pool:0xb1674700 SIP OnSetUp failed
for INVITE from sip:06814003340@127.0.0.1:6060 for
Call[C8e34455a2]-EPsip[b8e7402f-bab5-e211-972b-6c626db69c09]
2013/05/07 21:30:03.596OnRelease:0xb15b1700
ModemConnection::OnReleased Call[Cdd70c36e1]-EPmodem[modem:/Cdd70c36e1/0]
Call[Cdd70c36e1] cleared (Call cleared because the line is out of service)
2013/05/07 21:30:03.596OnRelease:0xb15b1700 Call[Cdd70c36e1] cleared
(EndedByOutOfService)
Call[C8e34455a2] cleared (Local party cleared call)
2013/05/07 21:30:03.695OnRelease:0xb15f2700 Call[C8e34455a2] cleared
(EndedByLocalUser)
2013/05/07 21:30:04.229 Opal Garbage:0xc71ba700 ttyT38-0 AudioEngine Attach
2013/05/07 21:30:04.229 Opal Garbage:0xc71ba700 ttyT38-0 AudioEngine
OnAttach Attached
2013/05/07 21:30:04.229 Opal Garbage:0xc71ba700 ttyT38-0 AudioEngine
OnResetModemState
2013/05/07 21:30:04.229 Opal Garbage:0xc71ba700 ttyT38-0 AudioEngine
ChangeModemClass to mcFax
2013/05/07 21:30:04.229 Opal Garbage:0xc71ba700 ttyT38-0 AudioEngine
OnChangeModemClass to mcFax
2013/05/07 21:30:04.229 Opal Garbage:0xc71ba700
ModemEngineBody::_AttachEngine Attached mceAudio
2013/05/07 21:30:04.229 ttyT38-0(e...0xb1778700 ttyT38-0 AudioEngine Detach
2013/05/07 21:30:04.229 ttyT38-0(e...0xb1778700 ttyT38-0 AudioEngine
OnChangeModemClass to mcUndefined
2013/05/07 21:30:04.229 ttyT38-0(e...0xb1778700 ttyT38-0 AudioEngine
OnDetach Detached
2013/05/07 21:30:04.229 ttyT38-0(e...0xb1778700 ttyT38-0 AudioEngine
OnResetModemState
2013/05/07 21:30:04.229 ttyT38-0(e...0xb1778700
ModemEngineBody::_DetachEngine Detached mceAudio
2013/05/07 21:30:04.229 ttyT38-0(e...0xb1778700 --  {
  0d 0a 42 55 53 59 0d 0a..BUSY.. }


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Re: [asterisk-users] Access postgresql directly from dialplan?

2013-04-17 Thread Sebastian Arcus

On 16/04/13 14:17, Gertjan Baarda wrote:

On 16 apr. 2013, at 15:08, Sebastian Arcus s...@open-t.co.uk wrote:


I would like to access a Postgresql database directly from my dialplan (to 
lookup names based on callerid numbers for incoming calls). Based on everywhere 
I looked - it seems the only way to do this is with func_odbc. Considering that 
Asterisk seems to be able to access Postgresql databases directly using 
res_pgsql for the purposes of realtime functionality (without using odbc - if I 
understand correctly) - is there some other generic SQL dialplan function which 
would allow access to the Postgresql database from the dialplan without 
configuring ODBC and the rest?

Or does anybody know of a simpler way still to do this? My database is a Horde 
address book backend - which is in Postgresql - so I would like to use it to 
convert the incoming callerid's into names on the fly from the Horde address 
book.



AGI is your new friend. Google it. I use it for your the same purpuses
as you want with PHP and Mysql.
With a macro I change the callerid in a name from the DB.

greetz



Thank you Gertjan. I have somehow managed to avoid AGI all these years 
and didn't even look into it to see what it does. I've just had a 
glance, and it seems to be exactly what I'm after. Thanks again for the 
suggestion


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[asterisk-users] Access postgresql directly from dialplan?

2013-04-16 Thread Sebastian Arcus
I would like to access a Postgresql database directly from my dialplan 
(to lookup names based on callerid numbers for incoming calls). Based on 
everywhere I looked - it seems the only way to do this is with 
func_odbc. Considering that Asterisk seems to be able to access 
Postgresql databases directly using res_pgsql for the purposes of 
realtime functionality (without using odbc - if I understand correctly) 
- is there some other generic SQL dialplan function which would allow 
access to the Postgresql database from the dialplan without configuring 
ODBC and the rest?


Or does anybody know of a simpler way still to do this? My database is a 
Horde address book backend - which is in Postgresql - so I would like to 
use it to convert the incoming callerid's into names on the fly from the 
Horde address book.


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Re: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation

2013-02-02 Thread Sebastian Arcus

On 01/02/13 09:43, Hans Witvliet wrote:

-Original Message-
From: Olivier oza_4...@yahoo.fr
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote
LAN workstation
Date: Thu, 31 Jan 2013 08:25:42 +0100

Hello,

On a LAN, is it possible to install a bluetooth dongle on one
workstation (at this time, this workstation OS is not specified) and use
it with chan_mobile ?
I've read some USB over IP (or Ethernet) middleware exist but I'm not
certain I'm looking at the right direction.

Regards
--
_
Hi Oliver,

I've been trying to do this for a while.
Been using latest blackberries and oldest nokia, and a laptop with
build-in and also an external BT-dongle.

What i noticed, is that the presence is highly unstable.
Even without walking along (kept the phone 10 cm from the BT-dongle) it
kept bouncing: found-gone-found-gone Unworkable.

I wanted not only presence, but speech-patch also via BT.
Idea was, that if co-workers are located at the other end of the world,
they can still be reached on they handy, even when no GSM-roaming is
acceptable (due to costs).

hw



Hi Hans,

As mentioned in my previous post - I've had this working for a number of 
years for my personal use. When I would go abroad, I would leave my 
cellphone/mobile phone home connected to Asterisk over bluetooth. I 
would say that if you're having problems staying connected - it might be 
the bluetooth dongle. It seems that the ones with the Cambridge Silicon 
chipset work best. In my case, once the phone gets connected, it stays 
connected. However, there is a known bug whereby Nokia phones 
automatically disconnect from bluetooth at the end of a call. To bypass 
this, I reduced the number of seconds to 5 in chan_mobile.conf - so that 
Asterisk tries to scan for the phone quick enough after it disconnects. 
I've added a note at the bottom of this page a while ago regarding this 
issue:


http://www.voip-info.org/wiki/view/chan_mobile

At the time I also got hold of as many phones as possible and tested 
compatibility. I've added my results to the same page above.


Sebastian

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Re: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation

2013-02-01 Thread Sebastian Arcus

On 31/01/13 10:15, Olivier wrote:



2013/1/31 Sebastian Arcus s...@open-t.co.uk mailto:s...@open-t.co.uk

On 31/01/13 07:25, Olivier wrote:

Hello,

On a LAN, is it possible to install a bluetooth dongle on one
workstation (at this time, this workstation OS is not specified)
and use
it with chan_mobile ?
I've read some USB over IP (or Ethernet) middleware exist but
I'm not
certain I'm looking at the right direction.

Regards



Hi Oliver,

I have used chan_mobile over the years on a number of occasions with
several different Nokia phones. I would say that even if in theory
it might be able to work with some USB over IP software for the
bluetooth dongle, it's probably not worth the hassle in practice.
It's quite likely that it will create too many problems, which will
probably outweigh the benefits of what you are trying to do.
Bluetooth already introduces a certain delay/latency in the
communication path - by adding and IP link in between, that will
only get worse.

Sebastian



Sebastian,

What I had in mind is to use someone's cellphone as a presence detector.
Let me explain:
- as the first thing you take along when leaving a room or location, is
your own cellphone, why not use chan_mobile and a bluetooth dongle on
your on PC (as you're not supposed to be within bluetooth range from an
asterisk server ;-)) to advertise you're away from your desk

- it seems that chan_mobile is not up to expectations for voice delivery
but would it remain the same for presence detection, if may call it this
way ?

Thoughts ?



Hi Oliver,

I'm am quite certain I read a bunch of material about using phones + 
bluetooth for presence detection. However, any particular reason you are 
trying to use Asterisk in this combination? I am quite sure there is 
software out there (I think for Windows and Linux) for bluetooth 
presence detection. What advantage would Asterisk bring for this scenario?


If you are trying to integrate it with Asterisk so that the phone 
switches to voicemail etc. - then maybe it is worth giving those 
usb-over-ethernet solutions a shot. After all, you are not trying to do 
voice over bluetooth - just detecting the mobile. I'm afraid I don't 
have any experience with that though. And I wonder what would the 
security implications be of having all those usb ports virtually 
connected to the server, but physically located away from the server?


Sebastian

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Re: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation

2013-01-31 Thread Sebastian Arcus

On 31/01/13 07:25, Olivier wrote:

Hello,

On a LAN, is it possible to install a bluetooth dongle on one
workstation (at this time, this workstation OS is not specified) and use
it with chan_mobile ?
I've read some USB over IP (or Ethernet) middleware exist but I'm not
certain I'm looking at the right direction.

Regards




Hi Oliver,

I have used chan_mobile over the years on a number of occasions with 
several different Nokia phones. I would say that even if in theory it 
might be able to work with some USB over IP software for the bluetooth 
dongle, it's probably not worth the hassle in practice. It's quite 
likely that it will create too many problems, which will probably 
outweigh the benefits of what you are trying to do. Bluetooth already 
introduces a certain delay/latency in the communication path - by adding 
and IP link in between, that will only get worse.


Sebastian


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Re: [asterisk-users] Is there a need to secure RTP ports?

2013-01-27 Thread Sebastian Arcus

On 25/01/13 12:31, Johan Wilfer wrote:

2013-01-23 18:20, Sebastian Arcus skrev:

I have an Asterisk server with one SIP trunk to a SIP provider. As my
server registers with the SIP provider, I don't have any SIP ports open
at my end to the Internet. However, I have the RTP ports open (as SIP
has some trouble with my NAT).


You could try iptables with ip_conntrack_sip ip_nat_sip.

If they are loaded and you accept calls from your sip provider on port
5060 iptables inspects the sip/sdp and traffic from the endpoints are
considered RELATED.

I've some research/testing to do myself on this topic (it's on my always
growing todo-list of doom.. :-)

Maybe you should check it out?


Thanks Johan. It will have to go on my todo-list of doom as well :-) - 
will eventually get around to investigating those two iptables options. 
Sounds promising.



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[asterisk-users] Is there a need to secure RTP ports?

2013-01-23 Thread Sebastian Arcus
I have an Asterisk server with one SIP trunk to a SIP provider. As my 
server registers with the SIP provider, I don't have any SIP ports open 
at my end to the Internet. However, I have the RTP ports open (as SIP 
has some trouble with my NAT). My question is - what are the 
vulnerabilities in this scenario at my end? I suppose some 
man-in-the-middle or eavesdropping  attack is always a possibility - but 
that aside, is there anything that will attack RTP ports on Asterisk 
when there are no SIP ports open? I was looking into installing fail2ban 
- until I realised that there is no SIP port exposed for an attacker to 
poke at.


Searching on Google for secure RTP ports keeps on bringing up results 
about SRTP - which is not exactly the answer to my question.


Thank you

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Re: [asterisk-users] Is there a need to secure RTP ports?

2013-01-23 Thread Sebastian Arcus
Thanks Danny. I've already reduced the number of RTP ports used in 
Asterisk configs and the firewall - as 1 seemed like a crazy number 
for my needs!


On 23/01/13 17:27, Danny Nicholas wrote:

As I am going to mis-explain this, an Asterisk SIP call originates on port
5060 (incoming or outgoing) then uses two RTP ports for audio in and audio
out.  Police and Hackers can tap into the RTP ports to monitor your
conversations (I don't really know if the capabilities stop there) but you
can limit your exposure by changing the default 1-2 range to a range
of 4 per anticipated calls simultaneously.  If you have 5 phones in your
shop, you aren't going to make 2500 simultaneous calls (just seems like
telemarketers can do this).  Change the 1-2 to 10001-10040 for a 5
phone shop.  This lets all 5 phones have two calls going at once.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
Arcus
Sent: Wednesday, January 23, 2013 11:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Is there a need to secure RTP ports?

I have an Asterisk server with one SIP trunk to a SIP provider. As my server
registers with the SIP provider, I don't have any SIP ports open at my end
to the Internet. However, I have the RTP ports open (as SIP has some trouble
with my NAT). My question is - what are the vulnerabilities in this scenario
at my end? I suppose some man-in-the-middle or eavesdropping  attack is
always a possibility - but that aside, is there anything that will attack
RTP ports on Asterisk when there are no SIP ports open? I was looking into
installing fail2ban
- until I realised that there is no SIP port exposed for an attacker to poke
at.

Searching on Google for secure RTP ports keeps on bringing up results
about SRTP - which is not exactly the answer to my question.

Thank you

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Re: [asterisk-users] Is there a need to secure RTP ports?

2013-01-23 Thread Sebastian Arcus

On 23/01/13 17:33, Carlos Alvarez wrote:

On Wed, Jan 23, 2013 at 10:20 AM, Sebastian Arcus s...@open-t.co.uk
mailto:s...@open-t.co.uk wrote:

I have an Asterisk server with one SIP trunk to a SIP provider. As
my server registers with the SIP provider, I don't have any SIP
ports open at my end to the Internet. However, I have the RTP ports
open (as SIP has some trouble with my NAT). My question is - what
are the vulnerabilities in this scenario at my end? I suppose some
man-in-the-middle or eavesdropping  attack is always a possibility -
but that aside, is there anything that will attack RTP ports on
Asterisk when there are no SIP ports open? I was looking into
installing fail2ban - until I realised that there is no SIP port
exposed for an attacker to poke at.


I've been working in IP telephony for about ten years.  I've never once
heard of any attack on the RTP ports.  While you can never say anything
is impossible there's simply nothing listening on those ports.  It's
probably possible to have a DOS attack where someone starts sending RTP
to all of your ports and they would interfere with a call, but they
couldn't do more than that.  That could work if your router has full
cone NAT and a lot of other things fall into place.  Still kind of out
there as a real threat.



Thanks Carlos. I sort of figured that there shouldn't be any listening 
daemons on RTP ports - as the calls get initiated on the SIP side of 
things - so couldn't think of any attack vector to compromise my server 
there. But I just didn't seem to be able to find an online source to 
back my theory. Thanks again for confirming.


Sebastian

Sebastian

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[asterisk-users] B200p card - use dahdi or mISDN?

2012-10-16 Thread Sebastian Arcus
I've just bought an OpenVOX B200p ISDN card - and if I remember 
correctly from last time I used one of these - it is possible to use 
either DAHDI or mISDN with it. Are there any factors to consider when 
choosing which software to use? Is one better than the other - or does 
one have features which are not present in the other?


I will be using it for a simple PBX, with 2 ISDN channels as trunks and 
9 SIP internal SIP extensions.


Or is this one of those questions where there are enough proponents on 
either side - so that there is no simple answer?


Thanks


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Re: [asterisk-users] B200p card - use dahdi or mISDN?

2012-10-16 Thread Sebastian Arcus

On 16/10/12 11:30, Patrick Lists wrote:

On 10/16/2012 08:50 AM, Sebastian Arcus wrote:

I've just bought an OpenVOX B200p ISDN card - and if I remember
correctly from last time I used one of these - it is possible to use
either DAHDI or mISDN with it. Are there any factors to consider when
choosing which software to use? Is one better than the other - or does
one have features which are not present in the other?


I would go for DAHDI so you can use the card like you would use any
Digium card. OpenVOX also seems to focus on DAHDI integration. Looking
at the OpenVOX site it seems that you will need to use the patched DAHDI
from here:

http://downloads.openvox.cn/pub/drivers/dahdi-linux-complete/


Thanks for that. I use Asterisk 10.x - it seems I only need the patch if 
using Asterisk 1.6?


Sebastian

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Re: [asterisk-users] B200p card - use dahdi or mISDN?

2012-10-16 Thread Sebastian Arcus

Thanks Andrew


On 16/10/12 12:08, Andrew Colin wrote:

I have worked with the B200P before and used the standard mISDN and the 
standard DAHDI and both worked fine.



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists
Sent: 16 October 2012 12:30 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] B200p card - use dahdi or mISDN?

On 10/16/2012 08:50 AM, Sebastian Arcus wrote:

I've just bought an OpenVOX B200p ISDN card - and if I remember
correctly from last time I used one of these - it is possible to use
either DAHDI or mISDN with it. Are there any factors to consider when
choosing which software to use? Is one better than the other - or does
one have features which are not present in the other?


I would go for DAHDI so you can use the card like you would use any Digium 
card. OpenVOX also seems to focus on DAHDI integration. Looking at the OpenVOX 
site it seems that you will need to use the patched DAHDI from here:

http://downloads.openvox.cn/pub/drivers/dahdi-linux-complete/

Regards,
Patrick


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[asterisk-users] Question about async channel or macro for monitoring a call

2012-09-25 Thread Sebastian Gutierrez
Hi,

 Im trying to do this:  


1) Originate a call between an external number and a ivr that do some things in 
background

2) after the originate I bridge the person that dial that extent with the 
external number

I would like to have the ivr in background while the bridge is up for 
monitoring porpoises, but seems to stop processing when the local bridge is done



other possibility could be having a Macro async??   when I make a dial and 
execute a macro I would like to put a while there and control some stuff, but 
until the macro is over I can´t have audio pass between the 2 channels that 
where dialled.


any hint on this??


thanks


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Re: [asterisk-users] chan_mobile

2012-09-18 Thread Sebastian Arcus

Hi Hans,

On 18/09/12 08:04, Hans Witvliet wrote:

Hi all,

In one of my other project i had a look at chan_mobile.
I build 1.8.15.1 with the apropiate module. (in my distro asterisk is
build without chan_mobile ;-)

After i filled in the mac-addresses of the BT-adapter and the one from
my phone, i see it is recognized, got connected, and immediate gets
disconnected.
What phone and bluetooth adapter are you using? Some of the most 
compatible phones for this sort of stuff seem to be Nokia phones. I've 
used three different models so far with success - although they always 
disconnect from bluetooth at the end of the call. They will reconnect 
again after a while. See note at the bottom of the page linked below.


When it comes to bluetooth adapters, the ones with Cambridge Silicon 
Radio (CSR) seem to be the best for this job.


The following page has some useful info:

http://www.voip-info.org/wiki/view/chan_mobile

Sebastian

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Re: [asterisk-users] Static noise on bridged calls to PSTN, although the trunk line is clean on its own

2012-09-13 Thread Sebastian Arcus

On 13/09/12 00:47, Vladimir Mikhelson wrote:


On 9/12/2012 5:33 PM, Sebastian Arcus wrote:

On 10/08/12 18:38, Chad Wallace wrote:

On Tue, 31 Jul 2012 09:44:26 +0100
Sebastian Arcuss...@open-t.co.uk wrote:


I have two setups with SIP hardware phones as extensions and POTS
lines as trunks. Internal SIP to SIP calls are crystal clear, but all
calls bridged to POTS have a significant amount of static noise. The
problem is that if I plug a POTS phone directly into the line, there
is almost no static noise - the line is clean. It's like Asterisk (or
the hardware) amplifies the static noise. What I've tried so far:

1. Connect Asterisk with a short cable directly into the master phone
socket, where it enters the building.
2. One of the lines carries ADSL - so I double filtered it.
3. Tried three different phone sets (one Grandstream, two Cisco
models).
4. Tried an OpenVox A400P PCI card and a Sangoma U100 USB
adapter as analogue-to-digital interfaces.


Have you run fxotune? I remember doing that when we had analog
lines. You'd have to look up how--maybe just in the fxotune man page.



Thanks for replying Chad - and sorry for the delay in my reply. I
should have mentioned that I ran fxotune and made no difference. I
also checked the interrupts, and even changed motherboard, and tried a
USB analog adapter (Sangoma U100) instead of the current OpenVox PCI
adapter. None solved the problem.

I have given in and asked the client to order ISDN lines I'm afraid.

Sebastian




Sebastian,

I understand it is too late for your client, but for the sake of
consistency.

I am experiencing a similar issue with Digium TDM410 on FXS lines. In my
case the static noise is always present on analog extensions provided by
TDM410.

I had a ticket opened with Digium, and they admitted the following, but
refused to make the findings public:

   1. The static noise is produced by Digium analog equipment on certain
  motherboards. I specifically tried various Dell Dimension Pentium
  III machines with several power supplies. They all consistently
  started producing the noise the moment DAHDI drivers loaded, even
  before Asterisk was loaded. When I tried Dell Dimension Pentium IV
  machine the noise was not there.
   2. On a machine which produced noise as described in #1 switching to
  a non-Digium TDM card fixed the noise problem. FXS daughter card
  stayed the same, just the base card was swapped.

I do understand your situation involved FXO, and switching to OpenVox or
Sangoma did not help. But I feel the root cause well may be the same.

Regards,
Vladimir




Thanks for sharing Vladimir. It is interesting that somebody else is 
experiencing at least similar symptoms as I am. The hardware is 
generating the static noise (or at least it amplifies it greatly) - but 
I just couldn't find any way to reduce it sufficiently. After a chat 
with somebody working for a telecom company which installs proprietary 
PBX's - I reach the conclusion that it is not worth the effort. He said 
that whenever they encounter noise on analog lines, they don't tend to 
waste time tuning and adjusting things. They just recommend switching to 
ISDN - so I figured it's the only reasonable thing left to do. Of 
course, this won't make up of the many, many unchargeable hours spent 
troubleshooting this :-)


Also, I'm afraid it isn't much of a solution for your case, if you are 
using FXS.


Good luck,

Sebastian

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Re: [asterisk-users] Trouble phoning via HUAWEI E169

2012-09-13 Thread Sebastian Arcus

On 13/09/12 11:16, Olivier wrote:



2012/9/13 Benedikt Schöffmann benedikt.schoeffm...@gmail.com
mailto:benedikt.schoeffm...@gmail.com

Hi there,

I'm setting up a Asterisk network and I ran into  some problems ...
as you might have guessed :)

The set up is like this:
Internal Communication in the company should be handled through
softphones over an asterisk server (works).
Outbound Communication should be handled through a HUAWEI E169
stick, accessed by the chan_dongle project.
http://code.google.com/p/asterisk-chan-dongle/

When I call internal numbers, everything works fine, but when I try
to access outside, I get the following error:
  == Using SIP RTP CoS mark 5
 -- Executing [06766770031@internal:1]
Answer(SIP/1001-0023, ) in new stack
 -- Executing [06766770031@internal:2] Dial(SIP/1001-0023,
dongle0/r1/06766770031,20,r) in new stack
[Sep 13 11:33:31] WARNING[9835]: channel.c:5603 ast_request: No
channel type registered for 'dongle0'
[Sep 13 11:33:31] WARNING[9835]: app_dial.c:2218 dial_exec_full:
Unable to create channel of type 'dongle0' (cause 66 - Channel not
implemented)
   == Everyone is busy/congested at this time (1:0/0/1)
 -- Executing [06766770031@internal:3]
Hangup(SIP/1001-0023, ) in new stack
   == Spawn extension (internal, 06766770031, 3) exited non-zero on
'SIP/1001-0023'

 From googling my way around, I know this type of error normally
relates to a module not being loaded, but chan_dongle.so shows up
when I type a module show. I've been fiddling around with this for
days and frankly I don't really know where the problem could lie.

Below are excerpts from sip.conf and extensions.conf

SIP.conf
code
[general]
bindport = 5060
bindaddr = 192.168.61.25
tcpbindaddr = 192.168.61.25
tcpenable = yes
context = internal
transport = udp
disallow = all
allow = gsm
allow = ulaw
allow = alaw

[dongle0]
type=friend
context=internal
audio=/dev/ttyUSB1
data=/dev/ttyUSB2
imei=359638011610601
imsi=232018830482446
transport=udp
disallow = all
allow = gsm
allow = ulaw
allow = alaw

[1000]
type=friend
callerid = Benny 1000
secret=1000
host=dynamic
canreinvite=no
dtmfmode=rfc2833
mailbox=1000
disallow=all
allow=gsm
allow=ulaw
allow=alaw
transport=udp
context=internal

[1001]
type=friend
callerid = Timme 1001
secret=1001
host=dynamic
canreinvite=no
dtmfmode=rfc2833
mailbox=1001
disallow=all
allow=gsm
allow=ulaw
allow=alaw
/code

Extensions.conf
code
[internal]
; for 4-digit numbers, assume it's a SIP number in our own context
; call it
exten = _,1,Answer()
exten = _,n,Dial(SIP/${EXTEN},20,r)
exten = _,n,Hangup

; else
; for a number starting with zero try to call via Dongle
exten = _0X.,1,Answer()
exten = _0X.,n,Dial(dongle0/r1/${EXTEN},20,r)
exten = _0x.,n,Hangup

/code

Please shed some light on this .

Kind regards,
Benedikt

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I've never tried chan_dongle, but to me, the Dial statement is incorrect.
Maybe the following would be better:

exten = _0X.,n,Dial(dongle/dongle0/r1/${EXTEN},20,r)




Looking at the chan_dongle documentation, it looks like you need to have 
a dongle.conf in /etc/asterisk. Do you have it and does it contain the 
right stuff? It looks like some of the stuff you've added to the 
sip.conf should really go in dongle.conf, at least according to this page:


http://wiki.e1550.mobi/doku.php?id=configuration

Actually, I'm not sure you should have any settings connected with the 
dongle in sip.conf - as SIP and dongle are different channel types and 
use different configuration files.


According to the examples on the same page, your Dial string should not 
include the name of the device, but the channel type, more like:


exten = _0X.,n,Dial(Dongle/r1/${EXTEN},20,r)


Also, what do you get when you run in Asterisk CLI:


dongle show devices

That should give you idea if the dongle is setup correctly in dongle.conf.

Hope the above helps,

Sebastian

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Re: [asterisk-users] Static noise on bridged calls to PSTN, although the trunk line is clean on its own

2012-09-12 Thread Sebastian Arcus

On 10/08/12 18:38, Chad Wallace wrote:

On Tue, 31 Jul 2012 09:44:26 +0100
Sebastian Arcuss...@open-t.co.uk  wrote:


I have two setups with SIP hardware phones as extensions and POTS
lines as trunks. Internal SIP to SIP calls are crystal clear, but all
calls bridged to POTS have a significant amount of static noise. The
problem is that if I plug a POTS phone directly into the line, there
is almost no static noise - the line is clean. It's like Asterisk (or
the hardware) amplifies the static noise. What I've tried so far:

1. Connect Asterisk with a short cable directly into the master phone
socket, where it enters the building.
2. One of the lines carries ADSL - so I double filtered it.
3. Tried three different phone sets (one Grandstream, two Cisco
models).
4. Tried an OpenVox A400P PCI card and a Sangoma U100 USB
adapter as analogue-to-digital interfaces.


Have you run fxotune?  I remember doing that when we had analog
lines.  You'd have to look up how--maybe just in the fxotune man page.


Thanks for replying Chad - and sorry for the delay in my reply. I should 
have mentioned that I ran fxotune and made no difference. I also checked 
the interrupts, and even changed motherboard, and tried a USB analog 
adapter (Sangoma U100) instead of the current OpenVox PCI adapter. None 
solved the problem.


I have given in and asked the client to order ISDN lines I'm afraid.

Sebastian


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[asterisk-users] Static noise on bridged calls to PSTN, although the trunk line is clean on its own

2012-07-31 Thread Sebastian Arcus
I have two setups with SIP hardware phones as extensions and POTS lines 
as trunks. Internal SIP to SIP calls are crystal clear, but all calls 
bridged to POTS have a significant amount of static noise. The problem 
is that if I plug a POTS phone directly into the line, there is almost 
no static noise - the line is clean. It's like Asterisk (or the 
hardware) amplifies the static noise. What I've tried so far:


1. Connect Asterisk with a short cable directly into the master phone 
socket, where it enters the building.

2. One of the lines carries ADSL - so I double filtered it.
3. Tried three different phone sets (one Grandstream, two Cisco models).
4. Tried an OpenVox A400P PCI card and a Sangoma U100 USB adapter as 
analogue-to-digital interfaces.
5. Reduced the software echo canceller in chan_dahdi.conf to 32 and even 
16 - until I could actually start to hear echo. Still no difference.
6. Reduced the rxgain and txgain in chan_dahdi.conf to 0 - but the 
static noise is still there.

7. Tried different phone cables for the pots line.
8. Tried a different motherboard on the computer with Asterisk and 
checked there is no IRQ sharing. Tried when there was no other load on 
the Asterisk computer.

9. Tried Asterisk 1.6, 1.8 and 10

Is there anything else I can do - or should I just give in to the static 
noise? Is that how other hybrid setups work - do you get static noise on 
the line - more than if plugged directly? The client is adamant that the 
noise on the line is too high - by comparison with the quality on mobile 
phone calls (which are digital, incidentally) - so if I don't find a 
solution, I suppose I will just have to rip it all out and let one of 
the companies with proprietary phone systems install one.


Any hints appreciated.

Sebastian

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[asterisk-users] audiohook errors

2012-05-28 Thread Sebastian Gutierrez
Hi,


I´m facing some issues on asterisk 1.8.10. I can see this on the console:

[May 28 15:46:19] ERROR[28099]: lock.c:438 __ast_pthread_mutex_unlock: 
audiohook.c line 705 (audio_audiohook_write_list): Error releasing mutex: 
Operation not permitted
[May 28 15:46:19] ERROR[28099]: lock.c:280 __ast_pthread_mutex_lock: 
audiohook.c line 688 (audio_audiohook_write_list): Error obtaining mutex: State 
not recoverable
[May 28 15:46:19] ERROR[28099]: lock.c:407 __ast_pthread_mutex_unlock: 
audiohook.c line 705 (audio_audiohook_write_list): mutex '(audiohook)-lock' 
freed more times than we've locked!


any ideas???


best regards

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Re: [asterisk-users] Fax Problem on direct FXO port

2012-05-18 Thread Sebastian Gutierrez
Hi Steve,

Thanks for the reply, I didn't change anything else, just changed spandsp 
version to de FFA, I use the spandsp version with more success in other places, 
but in this particular case, sending faxes works ok with both versions but with 
spandsp I couldn't receive any fax, with FFA I may get 70% of faxes ok.




On May 18, 2012, at 1:35 AM, Steve Underwood wrote:

 Hi Sebastian,
 
 has still some issues that not all faxes pass ok, but does the work == 
 still badly broken
 
 Your log doesn't seem to show a spandsp error. It looks more like a bad 
 signal. Did you change anything else when you installed FFA? Usually people 
 move the other way to improve their results.
 
 Steve
 
 
 On 05/18/2012 09:38 AM, Sebastian Gutierrez wrote:
 Rusty,
 
 thanks for the reply, the issue seems a spandsp issue, I changed to digium 
 free asterisk fax and works much better, has still some issues that not all 
 faxes pass ok, but does the work.
 
 thanks!
 
 
 
 On May 17, 2012, at 1:06 PM, Rusty Newton wrote:
 
 Sebastian,
 
 Seeing as this an issue related to faxing using the SpanDSP library; if you 
 do not get an answer leading to a solution here, then you may try asking on 
 the SpanDSP mailing list http://lists.soft-switch.org/mailman/listinfo
 
 It's likely that the Asterisk users, specifically using SpanDSP, may be on 
 that list.
 
 Thanks,
 
 Rusty Newton
 Open Source Community Support Manager
 Digium, Inc |www.digium.com  |www.asterisk.org
 
 On 5/16/2012 12:44 PM, Sebastian Gutierrez wrote:
 Hi,
 
 
 I´m with asterisk 1.6.2.20
 DAHDI Version: 2.5.0.2 Echo Canceller: HWEC, MG2
 SpanDSP: spandsp-0.0.6pre20.tgz 
 http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.6pre20.tgz
 
 FXO lines.
 
 Sending faxes works ok.
 
 but receiving gives me error:
 
 here is the debug:
 
 http://pastebin.com/qfFeXWQW
 
 
 any idea??
 
 
 Thanks!
 
 
 
 
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[asterisk-users] Fax Problem on direct FXO port

2012-05-16 Thread Sebastian Gutierrez
Hi,


I´m with asterisk 1.6.2.20
DAHDI Version: 2.5.0.2 Echo Canceller: HWEC, MG2
SpanDSP: spandsp-0.0.6pre20.tgz

FXO lines.

Sending faxes works ok.

but receiving gives me error: 

here is the debug:

http://pastebin.com/qfFeXWQW


any idea??


Thanks!

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[asterisk-users] 1.8 busypatterns

2012-05-07 Thread Sebastian Gutierrez
Hi,


is it possible to detect 4 length pattern busy cadence detection on FXO lines 
in 1.8??

Here the tones are:

425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off)


in asterisk 1.4 busy detect worked
in asterisk 1.6 didn´t work and i was told that 1.6 can´t handle 4 length 
patterns, but what about 1.8??

for now I can only hangup by asking the provider polarity switch.

Thanks

best regards.


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Re: [asterisk-users] 1.8 busypatterns

2012-05-07 Thread Sebastian
Can you point me to the commit to see if i can backport it?

Thanks
El 07/05/2012 18:50, Jonathan Rose jr...@digium.com escribió:

 - Original Message -
  From: Sebastian Gutierrez scg...@gmail.com
  To: asterisk-users@lists.digium.com
  Sent: Monday, May 7, 2012 10:38:03 AM
  Subject: [asterisk-users] 1.8 busypatterns
 
  Hi,
 
 
 
 
  is it possible to detect 4 length pattern busy cadence detection on
  FXO lines in 1.8??
 
 
  Here the tones are:
 
 
 
 
  425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off)
 
 
  in asterisk 1.4 busy detect worked
  in asterisk 1.6 didn´t work and i was told that 1.6 can´t handle 4
  length patterns, but what about 1.8??
 
 
  for now I can only hangup by asking the provider polarity switch.
 
 
  Thanks
 
 
  best regards.
 


 No. I implemented this in Asterisk 10.

 From the CHANGES file:

 from CHANGES in the trunk:

 --
 --- Functionality changes from Asterisk 1.8 to Asterisk 10
 ---

 --
 ...
 chan_dahdi
 --
  * Busy tone patterns featuring 2 silence and 2 tone lengths can now be
 used
   with busydetect.  usage example: busypattern=200,200,200,600

 So you'll need to upgrade to Asterisk 10 if you want to use that feature.

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Re: [asterisk-users] Line noise/hiss on Openvox A400P card on FXO

2012-03-13 Thread Sebastian Arcus

On 01/03/12 10:05, Sebastian Arcus wrote:

I have a server with an OpenVox A400P card with 2 FXO modules on it. The
internal extensions are SIP Grandstream phones. When making or receiving
external calls through PSTN, there is an interrupted hissing like high
pitch noise - which might go away for few seconds then start again.


Just a follow up to my own post. After taking apart the * server, 
replacing motherboard, replacing analog card with a usb FXO card from 
Sangoma - none of the above have helped with the problem.


However:

1. I tried disabling echo cancellation on the FXO ports in DAHDI and all 
of a sudden the line is much clearer. The noise is still there, but 
because the conversation is clearer, I could drop the gains on the FXO 
ports to -2db and -5db. This has reduced the background hiss to a 
certain extent. This was really an important lesson for me - as I would 
normally set the default echo cancellation (128 I think) and just leave 
it on - unless more was required. What I really should have done is 
started with no echo cancellation and just add a bit at a time - up to 
minimum necessary. I never realised how much harm too much echo 
cancellation does to the sound quality on the line.


2. I have tested the setup with an old Cisco 7940 phone. To my surprise, 
although the noise is still there somewhere, it is nowhere near as 
noticeable on the Grandstream GXP280 phone. It looks like in great part 
the Grandstream GXP 280 is just too sensitive to line noise - or it 
picks up / amplifies too much the wrong frequencies.


3. Again, using Ekiga, there is virtually no line noise. They must be 
using some really good algorithms which clean up the line.


Maybe the above will help someone. I just have to decide now if I scrap 
the Grandstreams and replace them with Cisco phones - or just live with 
the line quality.


Sebastian




1. The noise is not present when calling in between internal extensions
(SIP only).
2. The noise is the same on both PSTN lines.
3. The noise is NOT present when I tried two different phones directly
in the PSTN line(s) (a Philips DECT phone and a BT Converse phone)

Is the noise interference actually on the line, which the phones filter
out because of their better electronic design (then the OpenVox card) -
or is it generated somewhere in the server or on the OpenVox card?

I have tried:
1. Checking the interrupts and making sure the OpenVox card has its own
IRQ.
2. Moving the card around on different PCI slots.
3. Changing the second network card with a different model (the first
one is integrated in the motherboard).
4. Changing the motherboard, CPU and RAM (one motherboard AMD with Sis
chipset, the other one Intel).
5. Placing ferrite cores on the phone cables.
6. Checking to see if the OpenVox card gets 1000 interrupts per second
and it does.
7. Upgrading the kernel from 2.6.29 to 2.6.37
8. Ran FXO tune and made sure it starts with DAHDI
9. Disabled and enable software echo cancellation - it makes no difference.

The server is virtually under no load during the tests. It does have IDE
hard-drives (which apparently can cause problems) - but there is not
much I can do about that.

I also have a Sangoma USB FXO adapter - which I'm about to install and
configure to see if it makes a difference.

I would really like to figure out where is the noise coming from - as
I'm going a bit in circles. If I can find out for sure that the OpenVox
card is either broken or low quality - I'll just have to replace it. But
I can't even figure that out for sure.

The specs are:

CPU: Celeron 2.4GHz
Asterisk 10.1.2
Dahdi 2.6.0
Hard-drives: IDE
OpenVox A400P analog card



Many thanks for any advice.

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[asterisk-users] alaw = 1-2 in system.conf (dahdi) not working in UK

2012-03-11 Thread Sebastian Arcus

Hi all,

I've tried to explicitly set my two PSTN trunks/FXO lines to alaw with:

alaw = 1-2

in /etc/dahdi/system.conf. However, when I do this, all I get is loud 
intense noise on the line and nothing else - can't dial, can't make 
calls, can't receive calls. If I omit it altogether, everything works fine.


As I'm in the UK - I thought Europe uses alaw on phone lines - how come 
Asterisk seems to be using only ulaw - and alaw won't even work? Config 
examples on the Internet seem to suggest that it should work.


Thanks for any help,

Sebastian

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Re: [asterisk-users] alaw = 1-2 in system.conf (dahdi) not working in UK

2012-03-11 Thread Sebastian Arcus

On 11/03/12 14:07, Tzafrir Cohen wrote:

On Sun, Mar 11, 2012 at 12:05:48PM +, Sebastian Arcus wrote:

Hi all,

I've tried to explicitly set my two PSTN trunks/FXO lines to alaw with:

alaw = 1-2

in /etc/dahdi/system.conf. However, when I do this, all I get is
loud intense noise on the line and nothing else - can't dial, can't
make calls, can't receive calls. If I omit it altogether, everything
works fine.

As I'm in the UK - I thought Europe uses alaw on phone lines


No, they don't. Phone lines are analog. ulaw / alaw settings here are
internal to DAHDI. The digital PCM gets converted to analog audio
(voltage level) on the adapter.


- how
come Asterisk seems to be using only ulaw - and alaw won't even
work? Config examples on the Internet seem to suggest that it should
work.


Though it is, indeed, a bug.


Thank you Tzafrir. I initially though that the signal being analog, it 
can't be encoded with a digital compression algorithm - but as config 
examples online kept on suggesting the setting - I thought I would try it.





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Re: [asterisk-users] alaw = 1-2 in system.conf (dahdi) not working in UK

2012-03-11 Thread Sebastian Arcus

On 11/03/12 18:57, Shaun Ruffell wrote:

On Sun, Mar 11, 2012 at 06:49:01PM +, Sebastian Arcus wrote:

On 11/03/12 14:07, Tzafrir Cohen wrote:

On Sun, Mar 11, 2012 at 12:05:48PM +, Sebastian Arcus wrote:

Hi all,

I've tried to explicitly set my two PSTN trunks/FXO lines to alaw with:

alaw = 1-2

in /etc/dahdi/system.conf. However, when I do this, all I get is
loud intense noise on the line and nothing else - can't dial, can't
make calls, can't receive calls. If I omit it altogether, everything
works fine.

As I'm in the UK - I thought Europe uses alaw on phone lines


No, they don't. Phone lines are analog. ulaw / alaw settings here are
internal to DAHDI. The digital PCM gets converted to analog audio
(voltage level) on the adapter.


- how
come Asterisk seems to be using only ulaw - and alaw won't even
work? Config examples on the Internet seem to suggest that it should
work.


Though it is, indeed, a bug.


Thank you Tzafrir. I initially though that the signal being analog,
it can't be encoded with a digital compression algorithm - but as
config examples online kept on suggesting the setting - I thought I
would try it.


Hi Sebastien,

If for some reason you do need it to be in alaw, i.e. sometimes you
can get bettwe results when bridging to a digital line in alaw mode
if the audio is originally companded to alaw on the analog card, you
can use the 'companding' or 'alawoverride' module parameter
depending on the analog card that you're using.

There is an old open bug for this, DAHLIN-58 [1], which will require
a board driver / core of dahdi interface change to allow the board
drivers to be notified when the companding mode on a channel is
changed.

[1] 
https://issues.asterisk.org/jira/browse/DAHLIN-58?focusedCommentId=177136page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-177136

Cheers,
Shaun



Thanks Shaun. I'll keep this in mind in case it is ever needed.


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