Re: [asterisk-users] To Header instead of Request URI based routing

[Max Grobecker]

Hi,

do you have access to the system that sends you these calls?
If it's also an Asterisk, you could tell it to send another INVITE URI, 
regardless of what is submitted
in the registration.

On Asterisk with chan_sip you can do it by dialling:

 Dial(SIP/your_peer/+49202thatgoesinthetouri!+49202thatgoesintheinviteuri)

That is, as said, if the remote system which is sending you the calls is an 
Asterisk machine so you can
just reconfigure the way you get the calls to your local machine.

If it's not your system, you need to parse the To: header - for example, with:

 Set(ToHeaderVal=${SIP_HEADER(To)})
 Set(DailedNumber=${CUT(ToHeaderVal,:,2)})
 Set(DailedNumber=${CUT(DailedNumber,@,1)})

That should give you the dialed number in Variable "DialedNumber".


Greetings
 Max




Am 22.12.2017 um 14:54 schrieb Benoit Panizzon:
> Dear List
> 
> It looks like the common way to to sip signaling over a trunk is:
> 
> In the Request URI, return the 'Register' Contact.
> In the To: Header, send the destination number.
> 
> Unfortunately, asterisk with pjsip (i did not try chan_sip) does
> expect the dialed extension as request uri and does ignore what it is
> getting in the To: header.
> 
> I could not find any hint in the documentation of this can be changed.
> 
> I found instructions for a work-around:
> 
> http://www.kempgen.net/voip/sip-request-uri-vs-to-header-routing.html
> 
> In the meantime: Is there a way to tell the asterisk with pjsip to use
> the To: header to address an extension?
> 
> Kind regards
> 
> -Benoît Panizzon-
> 



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Re: [asterisk-users] Need to restart Asterisk if remote server not working?

[Max Grobecker]

Hi Luca,


Am 06.05.2017 um 15:49 schrieb Luca Bertoncello:
 
> I'm running an own BIND on my Linux-PC...

Me too ;-)


> Maybe should I configure a forwarder for the zone t-online.de? It not
> difficult, and if you mean it can help, I'll do that...

In the meantime, I setup forwarding requests to "t-online.de" and "t-ipnet.de" 
to the address 194.25.2.129.
That is kind of a global DNS resolver for all customers and is working since 
the 90s without address changes.



> Could you say me how can I disable the SRV lookups?
> I use Asterisk 1.8.30.0 on an OpenWRT device.

In your sip.conf, simply add

srvlookup = no

To your DTAG peer configuration.
If set globally, you may break the ability to directly call SIP addresses.



> The version of Asterisk on my OpenWRT unfortunately does not support dnsmgr...


On embedded systems, I often had problems with "stuck" DNS.
But that was ages ago... The last time on my old "Horstbox" with Asterisk 1.2 
and bristuff on Linux 2.4 :-/

Have you rebooted the whole WRT device or just restarted the Asterisk service 
to resolve your problem?
Maybe it's less an Asterisk issue but one with DNS caching on this device?



Viele Grüße aus dem Tal
 Max



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Re: [asterisk-users] Need to restart Asterisk if remote server not working?

[Max Grobecker]

Hello,

I'm also a customer of the DTAG.
Yesterday, the messed a bit with their DNS entries...

If you are NOT using their DNS resolvers you got a "wrong" IP address back that 
was not working.
Besides that, you should disable SRV lookups for their SIP peers. Since 
Asterisk's chan_sip.c does not honour the
weight of the SRV entries, nor it failovers to the other records, you might 
just end up with a not working server.
PJSIP might work with that, but it depends on your version.

The "blank" A record for "tel.t-online.de" is also provided and will be changed 
in case of service disruptions on one server,
so it's acceptable to rely on that.

DTAG is providing the following SIP servers at the moment (and also yesterday) 
with their SRV records:

_sip._udp.tel.t-online.de. 401  IN  SRV 0 5 5060 ims001.voip.t-ipnet.de.
_sip._udp.tel.t-online.de. 401  IN  SRV 1 5 5060 ims002.voip.t-ipnet.de.


ims001 should be the preferred one based on the SRV weight. But Asterisk only 
looks at the first record that comes as an answer,
so if ims002 is at the first position it will be used for registration, 
regardless that the other record is weighted better.
And if that one is not answering...
So: Better disable SRV lookups if you are not sure if your SIP channel driver 
supports it ;-)


You should also use the dnsmgr of Asterisk, resp. configuring it to reasonable 
values.
In dnsmgr.conf I set:

enable = yes
refreshinterval = 10


If dnsmgr is not enabled on your server this might have caused the problem 
because your SIP driver did not recognized that the
target address of the configured hosts has changed.
DNS changes should work also without dnsmgr - but since I've enabled the dnsmgr 
I had far less problems with changing DNS records ;-)



Am 06.05.2017 um 09:37 schrieb Luca Bertoncello:
> Hi list!
> 
> Yesterday Deutsche Telekom had a really big problem and Asterisk couldn't
> connect to the remote Server (by Telekom) until today about 7:30.
> 
> Well, it could happen...
> What I find really annoying was that I needed to restart Asterisk as I
> checked with sipsak that the Telekom-Server works...
> 
> I think, this should not be normal... Can someone explain me why it happens
> and what I have to change in the configuration to avoid this problem?
> 
> Thanks a lot
> Luca Bertoncello
> (lucab...@lucabert.de)
> 



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Re: [asterisk-users] How to have callers not being billed when in waiting queue ?

[Max Grobecker]

Hi,

in Germany, this kind of regulation is in effect for phone numbers which cost 
more than a normal landline call.
The regulation states, that the waiting time must not be charged to the 
customer.


Most companies implemented this by simply switching their telephone numbers to 
those, which are charged per call 
(so there's no difference in price between waiting for someone to pick up or 
being connected to someone) ;-)
Or they decided to use a normal landline phone number for which this regulation 
does not apply.

The second method was to not answer the call before really connected to a 
person on the queue and using Early Media as you mentioned.
But: The maximum length of this Early Media stream is in most telephone 
networks limited to somewhat around 90 to 180 seconds,
then the call gets disconnected by the network.

I'm not very familiar with regulations and numbering plans in France, but maybe 
there's also something called "offline billing".
Using this, your call is not billed by the caller's telephone company until you 
send them the amount of time that should be billed for a specific call.


Your best choice will be, that - if you ever get those regulations - you should 
rely on what your telephone number provider tells you to do ;-)


Greetings
 Max


Am 28.03.2017 um 15:24 schrieb Olivier:
> Hello,
> 
> In France, years ago, there was some discussions about a new regulation 
> forcing some providers to not charge anything to callers while those are 
> waiting for a call center agent to become available.
> Once caller and agent are on call with each other, nominal charging applies.
> 
> No matter if those discussions ever did or didn't change current regulation, 
> I wonder which dialplan statements could technically comply this dual billing 
> requirement ?
> 
> 
> same = n,Progress()
> same = n,Queue(whatever,...,macro-option, ...)
> 
> To me, coupling Progress app with Queue's  macro or gosub option like above, 
> would let a sysadmin answer a queued call.
> Doing so, time spent before connection with queue agent should not be billed 
> to anyone (caller nor callee), while time spent after connection is billed 
> normaly.
> 
> 1. Should this work ? Am I missing something ?
> 
> 2. Is there an alternative way to implement this ?
> 
> 3. Comments ? Suggestions ?
> 
> Regards
> 
> 



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Re: [asterisk-users] multiple outbound invites

[Max Grobecker]

Hi,

that could be caused when your upstream offers "100rel" and your Asterisk does 
not get a response fast enough from your upstream.
Is your outbound peer monitored by the qualify feature (qualify=yes)? 
Then asterisk should calculate the round-trip-time until a response arrives and 
should not resend the packets too fast.
If that does not help, you could play around with the "timert1" settings in 
your peer's SIP configuration.

Also, this sounds to me like a bug on the carriers side. It seems they are 
maybe offering "100rel" to you, but do not send any SIP/1xx responses
regarding your INVITE so your Asterisk resends these INVITEs because they are 
assumed lost.
Since these INVITEs all have the very same Call-ID and CSeq number, your 
carrier's equipment should be able to determine these packets are regarding
the same call. Again, I think your carrier should fix this problem on his site.
If he wants to enforce rate-limiting to INVITEs he should do it right by 
honouring the Call-IDs and sequence numbers.

If you like you can anyway send me your trace off-list, maybe there's something 
other weird going on.


Greetings
 Max



Am 22.02.2017 um 18:57 schrieb Jeff LaCoursiere:
> 
> Hello,
> 
> I have two upstream providers we use for US termination.  The dialplan sends 
> calls out the "primary" and if that fails for specific reasons, it sends the 
> same call out the "secondary". This has worked well for us when we are lazy 
> about keeping balances up, for example.
> 
> Starting a few days ago ALL calls sent to the 'primary' were returned as 
> busy, though the secondary terminated them fine.  We have a balance, and 
> funny enough international calls are going through fine, just not US calls.  
> I opened a ticket.
> 
> The response form the carrier is that our asterisk is sending four 
> simultaneous invites within one second, and for that reason the call is 
> rejected.
> 
> I did a packet trace and was able to confirm this is true - only US calls 
> sent to this carrier cause our end to send four identical simultaneous 
> invites.  When it fails, a single invite for the same call is sent to the 
> secondary, which is terminated without issue.
> 
> Happy to send the SIP trace if any would care to see it, but is there a 
> reason anyone can think of that our asterisk (11.11.0) would suddenly start 
> doing this?  It may be that it has been doing it all along, and our carrier 
> just started rejected calls that come in this way, I'm not sure.
> 
> Cheers,
> 
> j
> 
> 

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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Max Grobecker]

Am 16.02.2017 um 15:01 schrieb Joshua Colp:

> As for your issues please do file them. I'd also suggest using bundled
> PJSIP, it works the best with Asterisk and we backport applicable fixes
> and include fixes we've created that have not yet made it to a PJSIP
> release.

OK, I'll try again with the bundled version.
If the bugs persist, I'll file some bugs ;-)


Max



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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Max Grobecker]

Hi,

Am 16.02.2017 um 14:19 schrieb Annus Fictus:
> And Microsip using PJSIP SIP stack :)

Sorry (also, for off-topic), based on my latest experience with PJSIP, I'm not 
sure if this really is a sign of good quality.
Maybe it's a problem with the implementation in Asterisk (I haven't tried PJSIP 
in other software), but after just five minutes of testing
I found several bugs regarding PJSIP preventing me to use it in a production 
enviroment :-(
I'm going to file these bugs at the moment...


Max



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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Max Grobecker]

Hello,

I'm a big fan of PhonerLite.
It's more poplar in Germany, but also available in English language.
This client supports TLS, SRTP and ZRTP: http://phonerlite.de/features_en.htm

Yes, the GUI is not that much user friendly as Zoiper is - but at least a very 
good and stable client for testing purposes ;-)

Max


Am 15.02.2017 um 19:46 schrieb Motty Cruz:
> Hello, I have a user that prefers Soft SIP phone install on his laptop, for 
> security reasons I have enable TLS on our Asterisk server to support TLS 
> authentication, It works well with hard phones. Has anybody in this forum use 
> SIP Soft phones with TLS authentication enabled? Any suggestions?
> 
>  
> 
> Thanks,
> Motty
> 
> 
> 



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Re: [asterisk-users] SIP host name resolution

[Max Grobecker]

Hi,


Am 03.02.2017 um 18:23 schrieb Steve Edwards:
> If I have a SIP endpoint defined in sip.conf using a host name instead of an 
> IP address, do I have to reload sip to get Asterisk to 're-resolve' the host 
> name if I change the IP address in my DNS?

Normally, Asterisk honours DNS TTL and will re-lookup hosts as soon as the TTL 
is expired.
If you can't wait for that to happen, you can enable the builtin DNS manager 
and configure a refresh interval for DNS records to expire.
See "dnsmgr.conf" for the latter one.


> Does the answer change if the host name in sip.conf resolves to a CNAME and I 
> change the CNAME in my DNS?

Not as far as I know.
If you enabled SRV lookups for Asterisk, you may also want to check possibly 
existing SRV records for your host since Asterisk then looks them up first.


Max

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

[Max Grobecker]

Hi,

I figured out that this happens, when Asterisk ignores Session-Timers requests.
So I added the following to my DTAG peer configuration and eleminated the 
problem - and can use g722 on the DTAG network :-)

---
session-timers=accept
session-expires=120
auth_options_requests=yes
---


Greetings
 Max


Am 08.01.2017 um 19:47 schrieb Luca Bertoncello:
> Luca Bertoncello  schrieb:
> 
> Hi again!
> 
>> The problem: after 15 minutes will the call dropped, but only if the call is
>> to another nation! If I just call another phone in Germany, I can speak
>> longer than 15 minutes...
> 
> After a long work, and with the huge help of Michael Maier, I found the
> problem...
> I write here the description of the problem and my solution, maybe can this
> help someone other having the same problem...
> 
> The problem: after a successfully INVITE with the complete list of all
> supported Codecs, I receive about 15 minutes after call start, another INVITE
> (re-INVITE) from Telekom with __JUST__ one Codec: the one used by the call
> (currently: alaw).
> My Asterisk sends an "200 OK" with the same Codec and Telekom apparently has
> a problem with my answer, since the connection will be closed...
> 
> __MY__ solution: I configured Asterisk to use just __ONE__ Codec (alaw) for
> the communication with Deutsche Telekom.
> Now it seems to work, then I can call Italy and can speak longer than 15
> minutes.
> 
> I'm really puzzled and can't understand why Telekom has problem with my
> answer __JUST__ on calls outside Germany, but that is...
> 
> So, if someone other has the same problem, can try with my solution.
> 
> Hope to help!
> 
> Regards
> Luca Bertoncello
> (lucab...@lucabert.de)
> 

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Re: [asterisk-users] new inbound DID provider... no auth?

[Max Grobecker]

Hi,

That's right - you just need to define a peer with a static IP address and 
"type=peer" to assign incoming calls to a peer name and apply
the corresponding configuration (e.g. codecs).
To make your configuration less redundant you can use templates in your peer 
definition
(at least for chan_sip, I'm not sure if the same syntax applies on chan_pjsip).

Example:

---

;; All configuration made to this peer will be applied to all childs of this 
definition
[your-did-provider](!)
type=peer
allow=ulaw,alaw,g722
...

;; This peer derives all other configuration from "your-did-provider",
;; then your local changes are applied and can override the derived ones.
[your-did-provider-gw1](your-did-provider)
host=1.2.3.4

[your-did-provider-gw2](your-did-provider)
host=1.2.3.5
---

That's the shortest thing I can imagine at the moment.
At least, with this way of definition you only need to do changes on one single 
point, not for every gateway IP.





Am 30.11.2016 um 22:10 schrieb Jeff LaCoursiere:
> 
> We are trying to work with a new DID provider and I find myself confused.  
> Their standard integration is to send the call with no authentication.  I am 
> expected to whitelist all their possible gateways, and accept their calls I 
> guess with no peer definition.  I actually have it working this way; the 
> calls land in our "public" context, I guess as "guest", and I am able to 
> route them from there.  But that makes me nervous.
> 
> I would rather at least have them be associated with a defined peer, so I can 
> set the right context and any other parameters I might want associated.  It 
> is inbound only, no outbound.  I might try to set a host= in a peer 
> definition with no secret, and see if that matches it, but I would rather 
> avoid making a peer definition for every gateway they have.  Can anyone think 
> of a way to define a single peer that might show from multiple potential 
> addresses without authentication info?
> 
> Cheers,
> 



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Re: [asterisk-users] Change Media IP in SDP

[Max Grobecker]

Hi,

normally, Asterisk handles RTP IP addresses in SDP correctly, if you have 
specified
 - that NAT traversal is enabled for all peers (e.g. nat=force_rport,comedia)
 - your local network with "localnet=yournetwork/networkmask" - e.g. 
"localnet=192.168.1.0/255.255.255.0"
 - directmedia, canreinvite, directrtpsetup is deacitivated

In this case, your Asterisk will always stay in the RTP stream and signalling 
only it's own IP address to other peers
which is configured for the interface which Asterisk uses to reach the peer.


But: If you have only one interface configured on your Asterisk server and an 
external firewall/router is managing your separated networks, this might not 
help.
In this case you can use "externip" on a per peer basis in your SIP 
configuration to specify the IP address Asterisk uses in the SDP.
Maybe, a global configuration of "externip" and "localnet" is all you need to 
help Asterisk setting the SDP address correctly.
Also, enabling ICE support can help you getting the correct IP address if the 
remote peer supports it.



Greetings
 Max


Am 07.12.2016 um 00:02 schrieb Harel:
> Hello List,
> I need your help with information going out on my SDP.
> Is it possible to update the Media Address on a per-call basis or a 
> per-channel basis?
> Reason:
> My Asterisk is in a private network and needs to connect to UA on its 
> internal network and also few external networks. One network is public and 
> the others are not public. Between each other the external networks are not 
> routable. Signaling is flowing with no issues because SIP Registers and NAT 
> boxes maintain sessions correctly. The problem is with RTP. After making 
> traces on all possible nodes of this network I clearly found out that the RTP 
> fails because the Asterisk doesn't manage to communicate the correct address 
> to the UAs in the SDP. It will report its internal IP address and the remote 
> UA will try to send its RTP to this address which, of course, will fail 
> miserably. 
> Obviously I can't use externaddr or media_address in sip.conf because it will 
> only be good for one network while the other external networks will fail just 
> the same. Same applies for STUN, it will only be good for the network the 
> STUN requests are being sent from. 
> On all networks I have fix IP addresses on my side and I fully control a 
> professional security box. 
> Asterisk is 13.6.0
> I can't, and don't want to, touch user-side equipment which is normally some 
> kind of voip phone behind a standard home VDSL router.
> 
> Any ideas how can I transmit the correct IP address in SDP to UAs on 
> different networks?
> 
> Many thanks,
> Harel
> 
> 



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Re: [asterisk-users] Touch tone stutter

[Max Grobecker]

Hi,

you could try switching the DTMF mode of the ATA's SIP peer (and also in the 
ATA itself) to INBAND transmission.
In this mode, the ATA doesn't need to recognise DTMF tones and your Asterisk 
can interpret it.
For this to work, the ATA needs to use a G.711 codec. Inband DTMF needs an 
uncompressed codec to work properly.

Another way is (if the ATA supports it) to switch DMTF mode to SIP INFO.
In this mode, DTMF is not interpreted out of the audio stream. For external 
peers which are not supporting this mode
Asterisk then generates the proper RTP messages or tones.

With SIP INFO mode I made my best results with all devices, sadly it's not very 
common used.


Max


Am 23.11.2016 um 20:02 schrieb D'Arcy Cain:
> On 2016-11-22 07:49 PM, Pete Mundy wrote:
>>
>> One direction that may be worth exploring further is his ATA's config (or 
>> perhaps swapping it for a different model). Eg adjusting echo cancellation 
>> or line impedance settings.
> 
> I have to be careful here as I auto-provison these devices and changes would 
> propogate to every user.  Echo cancellation is off.  Do you think it should 
> be on?
> 
>> Is the ATA he is using the same as the ATA you use?
> 
> No but it is the same as other users who do not have the problem.  I use a 
> SIP phone and a Cisco ATA.
> 
>> Failure to correctly recognise and decode DTMF is just one of many reasons 
>> why I never use them (ATAs). Like faxing over VoIP, they're just too much 
>> trouble :(
> 
> I understand but some use cases just need it.
> 
>> Genuine IP phones are pretty good value these days. Could you drop one of 
>> those on-site as a temporary measure to prove that it's phone and/or ATA 
>> related?
> 
> He does want to have an extension so that won't work.
> 
>> Ps, you might also want to consider joining VoiceOps (if you're not already 
>> subscribed) and posting there. 
>> https://puck.nether.net/mailman/listinfo/voiceops
> 
> I have subscribed.  Thanks.
> 



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Re: [asterisk-users] Non-global variable that follows channel?

[Max Grobecker]

Hi,

is channel variable inheritance working for your setup?
Passing variables to other channels can normally simply be done by naming the 
variable with one or two prefixed undersorces
to make it available to the channel that is created from that one defining the 
variable.
But I have no idea if it's getting inherited to Gosub called from a Dial 
command...
 -> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Set

If that is not working for you, you might use the SHARED() variables which are 
kind of global accessible by the channel ID.
So you might call your Gosub with only the (unique) reference name of the 
variables you wish to pass and then call it from your Gosub.
 -> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_SHARED


Greetings,
 Max


Am 23.11.2016 um 13:06 schrieb Jonathan H:
> Related to 
> http://lists.digium.com/pipermail/asterisk-users/2016-November/290384.html,
> at the moment I'm passing one variable via DIAL.
> 
> Now I'd like to pass a whole bunch, and my idea was to rather than
> having a great string of
> 
> b(synctest3b^setVar^1(something)^2(more things)^3(etc))
> 
> and then get them with ARG1..ARGn etc, I could bundle the whole lot
> into a HASH and then unbundle them at the called channel.
> 
> Passing the HASH as a var isn't working (I wasn't expecting it to!)
> but is there any other way of doing this, or is it setVar for each
> one?
> 



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Re: [asterisk-users] Asterisk 11.24.1 garbled audio

[Max Grobecker]

Hi,

Am 17.11.2016 um 13:51 schrieb Jerry Geis:

> PBX Core settings
> -
>   Version: 11.24.1
>   Build Options:   LOADABLE_MODULES, BUILD_NATIVE
>   Maximum calls:   Not set
>   Maximum open file handles:   1024
>   Root console verbosity:  0
>   Current console verbosity:   5
>   Debug level: 0
>   Maximum load average:0.00
>   Minimum free memory: 0 MB
>   Startup time:16:23:00
>   Last reload time:16:23:00
>   System:  Linux/2.6.32-642.6.2.el6.x86_64 built by root 
> on x86_64 2016-10-30 20:40:02 UTC
>   System name:
>   Entity ID:   b0:83:fe:d1:af:5d
>   Default language:en
>   Language prefix: Enabled
>   User name and group: /
>   Executable includes: Disabled
>   Transcode via SLIN:  Enabled
>   Transmit silence during rec: Disabled
>   Generic PLC: Enabled
>   Min DTMF duration::  80



That's a bit odd... On my Asterisk 11 setup, I see an entry "Internal timing" 
which is totally missing on your installation.
You might want to try adding

  internal_timing = yes

to the [general] section of your asterisk.conf and then stop and start your 
Asterisk.

You can also try to unload all timing modules but "res_timing_timerfd.so" and 
try if it makes things better.
If it does, you can prevent res_timing_dahdi from being loaded in your 
modules.conf.


Max



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Re: [asterisk-users] Asterisk 11.24.1 garbled audio

[Max Grobecker]

Hi,


Am 15.11.2016 um 17:52 schrieb Olivier:
> Hi,
> 
> How can I double check which timer is currently is use in a running system ?
> core show settings doesn't tell anything, if I'm not mistaken.

To determine which timing module is currently in use, you can take a look at 
"module show like timing".
There should be only one module with "use count" 1 - that's the one that is 
currently used.
If there is no call running, you can unload any additional timing module you 
don't want to use to force Asterisk 
to use the only one left by simply doing "module unload res_".

Also, please check in "core show settings" if internal timing is enabled or 
not. If it's not, please enable it in asterisk.conf.
The internal timing should be enabled by default, but if it's not Asterisk 
might not use any timing module at all if RTP is being bridged between two ends 
of a call.
Asterisk normally synchronises the RTP clocking to one end of the call. But if 
this RTP source is not realiable (jitter, packet loss, silence suppression...) 
you
can end up having audio problems.



Max



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Re: [asterisk-users] SIP and RTP port and IP addresses

[Max Grobecker]

Hi Ethy,


Am 09.11.2016 um 17:13 schrieb Ethy H. Brito:

> How are these parameters available from dialplan?
> 
> For instance, ${SIPURI} holds the internal "IP:port" if the client is behind 
> NAT. 
> I need the external IP:port


You can get the peer's signalling IP address from ${CHANNEL(recvip)} and the 
RTP address with ${CHANNEL(rtpsource)} resp. ${CHANNEL(rtpdest)}.
If you need more information (like the codecs used) you can find other channel 
variables on 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CHANNEL

Please note that, if you have not disabled re-invites, the RTP address may 
change while the call is running.



Max



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Re: [asterisk-users] Suddenly getting lots of "Unable to send packet: Address Family mismatch between source/destination" but ONLY on 1 of 2 VPSs in same datacentre.

[Max Grobecker]

Hi Jonathan,

Am 05.11.2016 um 14:08 schrieb Jonathan H:

> What I don't understand is that while Ubuntu has IPv6 of course, the VPS host 
> is set to V6 disabled. and as far as I am aware, and my ITSP doesn't have 
> IPv6, so I just can't figure out why two IPv4 systems are getting IPv6 
> "pollution" as it were. And why now??!

That *MAY* be caused by a rogue IPv6 Routing Advertisement in the network where 
your vServer is located.
If you have a global IPv6 address assigned to your interface with the flag 
"dynamic" you got this address via autonomous addressing provided by routing 
advertisement.
To verify, look at the output of:

  sudo ip -6 addr show dev 

You'll find one or more lines starting with "inet6", followed by the assigned 
address and at the end of the line the flags;
For example "inet6 2003:..:1234/64 scope global dynamic" - this would be a 
dynamically assigned address.

Also, doing a

  sudo ip -6 route show default

Will bring more clarity, if you get a route entry like this:
"default via fe80::230:88ff:fe04:d dev ppp0  proto kernel  metric 1024  expires 
1539sec hoplimit 64"

The "expires" information indicates this route has been learned by RA. If you 
have no route entry this means you might have no IPv6 connectivity at all.
If there is a route entry but without "expires" information the route has been 
added manually.


If you have a global IPv6 address assigned to your interface, please check if 
it belongs to your providers network.
An easy way to check this is via https://stat.ripe.net (they use all RIR 
databases, so you'll find information about all regions).


In either way: Your provider should be worried about this.
Either there is a way for other customers to advertise (malicious) IPv6 routing 
information into the network that affects other customers
or your provider simply does not know that he is actively announcing and 
routing IPv6 or configuring customer's vServers with IPv6.

If it's a malicious or at least unknown advertisement, you definitely should 
deactivate the use of RA in your sysctl by setting in sysctl.conf:

  net.ipv6.conf.all.accept_ra=0
  net.ipv6.conf.default.accept_ra=0

Then, do a "sysctl -p" and manually remove the already assigned route.

The reason why you should not ignore this is:
When you get IPv6 routes via rogue advertisements and your servers is sending 
IPv6 traffic through the attackers server, he will be able to read your traffic.
And - for unencrypted VoIP traffic - he can simply see only all the numbers you 
dialed, seeing what DTMF keys were pressed and finally listen to the voice 
stream.

So - this is definetely worth to investigate and to get your ITSP have a look 
at it. There are many ways to stop other customers from doing this (maybe this 
happens accidently).
If you have further questions you might contact me off-list - since this is 
something that does not really fit in the asterisk list ;-)


Max



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Re: [asterisk-users] Just got defrauded - how do I block calls which contain a dash (RegEx noob question)

[Max Grobecker]

Hi,


Am 28.10.2016 um 17:38 schrieb Markus:
> exten => _-.,1,NoOp(Blocking dash)
> exten => _-.,n,Hangup

> How do I do it right?


why not using FILTER() in your dialplan to eleminate all chars that are not 
numeric?
Like 

   Set(VAR=${FILTER(0-9+),${EXTEN}})

That would eleminate all characters you're not expecting.


Greetings
 Max



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Re: [asterisk-users] Adding a pause when transfering a call

[Max Grobecker]

Hi,

some phones can add a pause when dialing, sometimes by holding the * or # key a 
few seconds after the first digit.
If it works, the phone normally adds a "W" or ";" to the dial string.
So you would program the speed dial key with <*2[hold * or #]101>.

Am 01.10.2016 um 20:22 schrieb Tech Support:
> All;
> 
> When I transfer a call to another extension, I can simply press *2 and 
> then the extension number, say 101. No big deal. The problem I am having is 
> in programming a speed dial key to dial *2101, which is failing. The only 
> thing I can think of is that the speed dial key is dialing the string too 
> fast and Asterisk sees it as <*2101> instead of <*2><101> which fails. How do 
> other people get around this?
> 
> Thanks;
> 
> John  
> 
> 
> 



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Re: [asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

[Max Grobecker]

Hi,

OK, then it looks like the client transferred the call anywhere else.
Do you see an entry in your log that refers to the bridge ID 
00bd58c3-3bce-4f1b-9d79-11eb96f37260 ?
If there was a transfer, the call *may* have been bridged with the transfer 
destination. Also, the destination might be external,
so you may see a second call starting at the time where the client left the 
bridge.

Max



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Re: [asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

[Max Grobecker]

Maybe the client just put the call on hold.
So the call technically has not ended AND the client does not need to
send or handle any RTP data.
Is there any mention of "music on hold" for this channel?

Greetings
 Max


- Nachricht von Leandro Dardini  -
 Datum: Thu, 15 Sep 2016 18:06:14 +0200
   Von: Leandro Dardini 
Antwort an: Asterisk Users Mailing List - Non-Commercial Discussion

   Betreff: [asterisk-users] Tricking asterisk to think the call has
ended, but it was continuing on the other side
An: Asterisk Users Mailing List - Non-Commercial Discussion




I am banging my head over a simple asterisk trick I was seeing on one
asterisk server.

An extension dials an international premium number, the called number
answers, then the extension hangups, but the call continue to run on the
international number side, generating an high profit for the premium number
company and a big loss for the asterisk owner.

I think some sort of "transfer" takes place, but I can't identify how they
do it and most important, how to prevent it.


- Ende der Nachricht von Leandro Dardini  -




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[asterisk-users] Get Realtime extension matched entry ID

[Max Grobecker]

Hello,

is there a possibility to get (by dialplan variable?) the entry ID of
the realtime extensions table,
that matched the current call?

For example (simplified):

ID - exten
-
1+49123456
2_+49555.


If I receive a call on +49123456 this surely works with REALTIME_FIELD
and ${EXTEN} as matching field value.
But for ID 2 in ${EXTEN} the full dialed number is stored, so I would
never find a matching field in the database using this way.


Is there any way to get the ID field of the current channel or at
least a variable, where the unexpanded matched "exten" pattern is
stored (i.e. the "_+49555.")?
I just need something unique to find the dialed extension in the table...


Thanks!

Greetings,
 Max


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Re: [asterisk-users] Asterisk 13.11 realtime problem registering phones

[Max Grobecker]

Hi,


Am 02.09.2016 um 22:48 schrieb Carlos Chavez:
> I upgraded my office installation from 13.10 to 13.11 yesterday and now I 
> am having problems registering phones.  Here is what I get on the CLI:
> 
> [Sep  2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: 
> Realtime table general@ps_contacts: column 'qualify_timeout' cannot be type 
> 'int(10)' (need char)
> [Sep  2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: 
> Realtime table general@ps_contacts: column 'expiration_time' cannot be type 
> 'bigint(20)' (need char)
> [Sep  2 15:38:46] WARNING[2098]: res_config_mysql.c:1246 require_mysql: 
> Possibly unsupported column type 'enum('yes','no')' on column 
> 'authenticate_qualify'
> [Sep  2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: 
> Realtime table general@ps_contacts: column 'via_port' cannot be type 
> 'int(11)' (need char)
> [Sep  2 15:38:46] ERROR[2098]: res_pjsip_registrar.c:411 register_aor_core: 
> Unable to bind contact 
> 'sip:2001@192.168.2.203:57776;transport=UDP;rinstance=48d5c7d09b9f2525' to 
> AOR '2001'
>   == Contact 
> 2001/sip:2001@192.168.2.203:57776;transport=UDP;rinstance=48d5c7d09b9f2525 
> has been deleted
> 
> The mysql warnings have always been there since version 13.0 and the 
> "Unable to bind contact..." error has also been present since I started using 
> PJSIP realtime with Asterisk 13 (13.5 at least). 

I hope you find this concerning...

Have you upgraded your MySQL realtime tables to the new schema as introduced 
with Asterisk 13?
-> 
https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13#UpgradingtoAsterisk13-RealTime

It's likely a database error (i.e. a required, but missing table field) causes 
this issue.
But even if not, you are getting rid of the warning messages ;-)



Greetings
Max



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Re: [asterisk-users] Trouble getting peer variable (sip username) on 302 Moved Temporarily

[Max Grobecker]

Hi Jonas,


Am 02.09.2016 um 11:26 schrieb Jonas Kellens:
> [Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back from 
> 11.22.33.44:40670
> [Aug 31 14:59:34] -- Now forwarding 
> Local/myaccount184@CallFromQueue-07f4;2 to 'Local/23@from-internal' 
> (thanks to SIP/myaccount184-3729)
> Question : how can I read the variable which contains the value 
> 'myaccount184' in the context from-internal ?

You can get some information out of the REDIRECTING function [1].
For example, your redirecting source (the called device that caused call 
diversion) is normally stored in REDIRECTING(from-num).

[1] https://wiki.asterisk.org/wiki/display/AST/Function_REDIRECTING


Greetings
Max



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