Re: [asterisk-users] Global variables in global variables

[Daniel]

Le 25/01/2023 à 17:56, Antony Stone a écrit :

On Wednesday 25 January 2023 at 16:46:14, Daniel wrote:

On Sunday 01 January 2023 at 17:30:03, Antony Stone wrote:


The [globals] section of that dialplan includes:

Kphones=SIP/KC470IP/KSnom870
Sphones=SIP/SYealinkT38G/SGC610IP
Allphones=${Kphones}&${Sphones}

On the new system, the variable Allphones ends up containing:

${Kphones}&${Sphones}


I do the same concatenation with Asterisk 18 & 20 and there is no problem.


Really?

You have something like:

Allphones=${Kphones}&${Sphones}

and specifically *in the [globals] section* of the dialplan?


Asterisk 20.1.0

[globals]
Sphones=SIP/SYealinkT38G/SGC610IP 



Kphones=SIP/KC470IP/KSnom870 



Allphones=${Sphones}&${Kphones}

-s*CLI> dialplan show globals
Allphones=SIP/KC470IP/KSnom870/SYealinkT38G/SGC610IP
Sphones=SIP/SYealinkT38G/SGC610IP
Kphones=SIP/KC470IP/KSnom870

0
--
Daniel Huhardeaux
+33.368460...@tootai.net  sip:8...@sip.tootai.net
+41.445532...@swiss-itech.chtootaiNET

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Re: [asterisk-users] Global variables in global variables

[Daniel]

Le 25/01/2023 à 11:06, Antony Stone a écrit :
[...]


I await the repair of whatever has been delaying messages on the list, and
then I am optimistic that someone will have replied, even if it takes some
days for that reply to become apparent.


I do the same concatenation with Asterisk 18 & 20 and there is no problem.

BTW you should move to asterisk community lots more people there.
--
Daniel

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Re: [asterisk-users] Delay when dialing...

[Daniel Van Den Berg]

Hi there,

I can confirm that this is indeed the problem.

If you follow the advise below you will be sorted.

⁣From my mobile phone​

On 23 Jul 2021, 8:44 am, at 8:44 am, Jean Aunis  wrote:
>Le 22/07/2021 à 18:32, Carlos Chavez a écrit :
>>     I started noticing a few days ago that whenever I dial any number
>
>> or extension there is a delay of 5 to 10 seconds before Asterisk
>> reacts.  I see nothing on the CLI for that time and then the call
>goes
>> through.  I have checked my network to make sure there is nothing
>> slowing down packets between the phones and the server.
>>
>>     Any settings I should check on the Asterisk side?  This is
>> happening with all phones (several brands).
>>
>Hi,
>
>I've seen this problem several times when there is no DNS resolution of
>
>Asterisk's hostname.
>
>Try to add your hostname to /etc/hosts and check if it's better.
>
>Regards,
>
>Jean
>
>
>
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[asterisk-users] ARI strange bug on version 13.29.2

[Daniel Friedman]

Hello,

I am using an ARI dialer for my applications and since my last upgrade
to Ver. 13.29.2 from 13.23.1 I am getting this strange bug from the ARI 
debugger:

Debugging on all applications enabled
<--- ARI request received from: x.x.x.x:63036 --->
HOST: x.x.x.x:8088
content-type: application/json
authorization: Basic 
content-length: 265
body:
{
  "context": "from-itc-dialer",
  "extension": "secondleg",
  "priority": 1,
  "timeout": 60,
  "endpoint": "Local/firstleg@from-itc-dialer/n",
  "variables": {
"AGENT": "506655579",
"DESTINATION": "1866222",
"CLI": "442031502032"
  }
}

[2019-12-17 02:10:06] ERROR[24851]: json.c:870 ast_json_vpack: Error building 
JSON from '{s: s?, s: s?, s: o, s: s?, s: s?}': Expected format 's', got '?'.
[2019-12-17 02:10:06] NOTICE[24853][C-208b]: Ext. firstleg:5 @ 
from-itc-dialer: The group channel is: itc-ebay and the group count is: 1 out 
of max 14
[2019-12-17 02:10:06] ERROR[24851]:   Got 16 backtrace records
# 0: asterisk ast_json_vpack()
# 1: asterisk ast_json_pack()
# 2: asterisk ast_json_dialplan_cep_app()
# 3: asterisk ast_channel_snapshot_to_json()
# 4: res_ari_channels.so ()
# 5: res_ari_channels.so ()
# 6: res_ari_channels.so ()
# 7: [0x7f71a7385098] res_ari.so :0 ast_ari_invoke()
# 8: [0x7f71a73865e1] res_ari.so :0 ast_ari_json_format()
# 9: asterisk ()
#10: asterisk ()
#11: asterisk ()
#12: asterisk ()
#13: asterisk ()
#14: [0x7f71e489caa1] libpthread.so.0 :0 pthread_create()
#15: [0x7f71e3c2493d] libc.so.6 :0 __clone()

<--- Sending ARI response to x.x.x.x:63036 --->
200 OK

The calls are getting through and the Asterisk is not crashing.
I looked at the source code and it seems that I am sending all the right 
parameters with a valid JSON format.

Anyone has an idea what has been change in the Asterisk code and how can I 
adopt my code to the this version?


Regards,

Daniel Friedman
Trixton LTD.

Tel: 972.72.2557000
Mobile: 972.50.6655579

Email: d...@3xton.com<mailto:d...@3xton.com>
Website: http://www.3xton.com

[LOGO trans- 3xton-01]


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Re: [asterisk-users] Is there any way to pass caller id to cell phone

[Daniel Friedman]

Hello,

You can ask your provider to accept PAI headers that you 
Would add to your SIP Invite request.

Usually, this is what you do when you want to block
Your caller id from showing it to the callee.
The only way that the provider can identify you (for billing and legal purposes)
Is by RPID or the PAI headers.

Regards,

Daniel Friedman
Trixton LTD.

Tel: 972.72.2557000
Mobile: 972.50.6655579

Email: d...@3xton.com
Website: http://www.3xton.com





-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of asterisk-users-requ...@lists.digium.com
Sent: Monday, October 15, 2018 8:00 PM
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Subject: asterisk-users Digest, Vol 170, Issue 17

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Today's Topics:

   1. Re: asterisk 16 manager --END COMMAND-- (Jacek Konieczny)
   2. Re: Is there any way to pass caller id to cellphone? (Eric Klein)


--

Message: 1
Date: Mon, 15 Oct 2018 08:36:23 +0200
From: Jacek Konieczny 
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk 16 manager --END COMMAND--
Message-ID: <44e25e43-e845-92f6-4e56-8e67f8643...@jajcus.net>
Content-Type: text/plain; charset=utf-8

On 2018-10-12 12:22, Dmitry Melekhov wrote:

>> AMI:
>>   - The Command action now sends the output from the CLI command as a 
>> series
>>     of Output headers for each line instead of as a block of text 
>> with the
>>     --END COMMAND-- delimiter to match the output from other actions.
>>
>>     Commands that fail to execute (no such command, invalid syntax
>> etc.) now
>>     return an Error response instead of Success.
>>
> Very pity that you break compatibility...

The old AMI protocol was so broken, so it was hardly possible to make any 
compatible client implementation. Whatever you do, it would break on some 
corner cases. This change fixed a little bit of this mess.

And if some client library is not properly updated for major Asterisk releases, 
then that is not Asterisk to blame.

Jacek



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Message: 2
Date: Mon, 15 Oct 2018 11:12:09 +0300
From: Eric Klein 
To: asterisk-users 
Subject: Re: [asterisk-users] Is there any way to pass caller id to
cellphone?
Message-ID:

Content-Type: text/plain; charset="utf-8"

Ivan,

Be aware that what you are asking may cause problems with making the call to 
the cell phone.
Think of it this way, you are taking an inbound call and then sending it out 
over your regular operator. They may object to accepting a call with a CLID 
that does not match your account and could block it.
It is worth testing if they will allow any outbound CLID or need it to match 
the account. The problem will get worse when SHAKEN'STIR comes into effect and 
they need to certify that the call came from your office.
The reason they would block it is to prevent both spam calls and fraud.

Eric Klein
COO
Greenfield
Main US +1 805 410 1010
Main UK +44 203 746 6000
Main Il+972  73 255 7799
Mobile+972 54 666 0933

*Email *e...@greenfield.tech
Skype: EricLKlein
Web: www.greenfield.tech
 www.cloudonix.io


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> Date: Thu, 11 Oct 2018 17:18:24 + (UTC)
> From: Ivan Demkovitch 
> To: "asterisk-users@lists.digium.com"
> 
> Subject: [asterisk-users] Is there any way to pass caller id to cell
> phone?
> Message-ID: <1490413779.8332018.1539278304...@mail.yahoo.com>
> Content-Type: text/plain; charset="utf-8"
>
>
> We have following problem. On some of the extentions I call cell phone 
> after 10 seconds or so.Or, like this o

Re: [asterisk-users] Is there any way to pass caller id to cell phone?

[Daniel Tryba]

On Thu, Oct 11, 2018 at 05:18:24PM +, Ivan Demkovitch wrote:
> Where problem comes in - if person not at the desk - his cell phone shows 
> call from OFFICE number and there is no way to tell who is really calling.
> We use Callcentric as a trunk if it makes any difference.
> I'd like to add info about caller when passing to cell phone if possible. Is 
> there any way to do that?

Maybe you should ask them how to do this! Maybe you should add a
Diversion header, maybe they don't allow this kind of spoofing at all.
This is a common request from users of SIP trunks and your use case is
legit. If Callcentric does checks on callerid validity and there is a
call to a customer with callerid X, they should be able to use this
callerid X when forwarding to an external device/number.


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Re: [asterisk-users] Disabling a trunk at runtime

[Daniel Tryba]

On Fri, Oct 12, 2018 at 07:59:52AM -0400, Telium Support Group wrote:
> I have an Asterisk system with 2 trunks (as shown below).  I need to be able
> to disable a trunk at runtime. I may not change the dialplan but I can
> change sip.conf and reload.
> 
> Any attempt to dial in the dialplan uses trunk A and trunk B in that order.
> Normally calls will route through trunk A, but if I disable A I want calls
> to go to trunk B.
> 
> Is there a creative way to effectively disable a trunk at runtime given
> these parameters?  I don't think there is an "enabled" key-value pair for
> sip.conf stanzas.  If I change the host key value to 0.0.0.0 and reload will
> that effectively cause the dialplan to use trunk B?
[snip]

TIMTOWTDI:

- You can create a dialplan that checks a global variable whether to skip
  trunk A. You can manipulate this variable from the AMI.

- Use an AGI script to set variables or dial trunks directly.

- use a script to generate configuration (included files) and reload the
  channel driver on changes.

- Do (no)sql queries from the dialplan.

- And probably lots more of possibilities.

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Re: [asterisk-users] 401 unauthorized

[Daniel Tryba]

On Wed, Aug 29, 2018 at 11:37:34AM -0400, Jerry Geis wrote:
> I have a connection to a cisco all manager SIP trunk. The first call coming
> across CCM to the asterisk server works fine... Then when I do a second
> call from CCM to asterisk  I am getting a SIP 401 unauthorized.
> 
> My definition is simple.
> [CCM]
[no secret]
> 
> This is asterisk 13.19.0
> 
> What is wrong? Thanks

The reason why the CM is asking for authentication is outside the scope
of asterisk, debug the CM. Since you have no secret set for the user,
asterisk will never try to respond to a challenge. Asterisk and CM can
work together, asterisk is very forgiving, CM is not. The last time I
had to figure out what the fromuser and fromdomain should be (the CM admins
were clueless about providing those facts) and there was some
throtteling mechanism that when triggered would deny any SIP messages.

Good luck.

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Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

[Daniel Journo]

> It does seem like a bug.  However, you have a complicated dialplan with a lot 
> of pieces happening at
> once so it may not actually be an Asterisk bug but a problem with your 
> dialplan.  To unravel this is
> going to take some bookkeeping on your part.

Hi Richard,

Thanks for the detailed response. Need to get my head around it a bit!

I’m going to try to set up a test rig with a less complex dialplan.
I’ll then run some tests and will be able to supply sample files if the problem 
persists.

I’m just a little confused what the different is which transfers of inbound 
queued calls and transfers of inbound Dialled calls.
I wonder if it would make a difference if the queue members were 
Local/DialThisEndpoint_200, instead of PJSIP/endpoint_200.

Since the queue dials the members and the member channels inherit the variables 
correctly, that would mean that Local/DialThisEndpoint_200 would inherit the 
DYNAMIC_FEATURES.
The Local channel would dial the endpoint, and when the endpoint performs a 
transfer and loses its variables, the Local channel, as the parent, would still 
have its variables set and the feature codes would still work.

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Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

[Daniel Journo]

> Doing some more tests, this reads like a bug to me.
> Using a hanguphandler with DumpChan in the dialplan context that executes
> the Queue, I can see that DYNAMIC_FEATURES is set.
> After the attended transfer when the call is ended, the hanguphandler still
> shows that DYNAMIC_FEATURES is set. It's just not accessible.
>
> Any thoughts?
> It likely depens on how you are doing the attended transfer.  Via DTMF?  Via 
> SIP
> or channel technology protocol?
> Does the Agent B channel have the DYNAMIC_FEATURES channel variable set
> on it?
>

Thanks for the reply.

To answer your question, the attended transfers are done via the endpoint's 
feature buttons. So I assume it's via SIP requests.

I've been doing some tests and reviewing the debug logs to try to understand 
the problem and still think it's a bug at this point.

Firstly, most of my inbound calls are answered and then Dial() Local channels. 
These Local channels set __DYNAMIC_FEATURES and various other things. They are 
also needed to ensure functionality like MixMonitor can be started on the Local 
channel and then not affected by any transfers. The Local channels then either 
Dial() some peers via other local channels (as some peers are required to press 
1 to accept the call) or a Local channel that dials a Queue().

For non-Queue calls that are going via the Local Channels that only use Dial().
When endpoint_201 dials *1, it is matched with their own channel.
> DTMF feature hook 0x7f18d803a978 matched DTMF string '*1' on 
> 0x7f18c000b080(PJSIP/endpoint_201-cb55)

Interestingly, after the attended transfer from endpoint_201 to endpoint_202, 
when endpoint_202 dials *1, it can no longer match and passes it back to the 
Local channel that originally dialled endpoint_201.
At that point, it can match the local channel since that's where 
DYNAMIC_FEATURES was originally set.
> No DTMF feature hooks on 0x7f189c0660b0(PJSIP/endpoint_202-cb5b) match 
> '*1'
> DTMF feature hook 0x7f1894abd408 matched DTMF string '*1' on 
> 0x7f186c04efa0(Local/fromfeature_201@phones-5a17;1)
So although the transfer caused the variable to be lost, the Local channel, as 
the parent, remained and stepped in to complete the *1 request.
Probably works by accident.

But calls passing through a Local channel that ends in Queue() don't act the 
same way.

The Queue's initial dial of the queuemembers includes the inheritance as 
expected.
So when endpoint_201 answers and they dial *1, this is the result.
> DTMF feature hook 0x7f18c4018278 matched DTMF string '*1' on 
> 0x7f18b0002ca0(PJSIP/endpoint_201-cacc)

But following a transfer, using the same SIP messaging as the non-queue calls, 
this is the result...
> DTMF feature string on 0x7f18bd369720(PJSIP/endpoint_202-cae8) is now '*1'
> No DTMF feature hooks on 0x7f18bd369720(PJSIP/endpoint_202-cae8) match 
> '*1'
> Playing DTMF stream '*1' out to 
> 0x7f18e4112980(Local/queue_dialplan_101@queue-59b2;2) < this channel 
> still has DYNAMIC_FEATURES set (see below) but it just passes the DTMF 
> through?
> DTMF begin '*' received on Local/queue_dialplan_101@queue-59b2;1
> DTMF begin passthrough '*' on Local/queue_dialplan_101@queue-59b2;1
and it's passed all the way back and played to the caller.

This is in spite of the fact that Local/queue_dialplan_101@queue-59b2;2 has 
DYNAMIC_FEATURES set earlier in the dialplan.
> Set("Local/queue_dialplan_101@queue-59b2;2", 
> "__DYNAMIC_FEATURES=NewRecordApp")
And still set at the end of the call, confirmed using DumpChan within the 
channels hangup handler.

> Dumping Info For Channel: Local/queue_dialplan 101@queue-59b2;2:
> Variables:
> DYNAMIC_FEATURES=NewRecordApp

I can't really explain why the channel can still have DYNAMIC_FEATURES, but 
it's not perform matching apart from thinking it's a bug.

I hope that wasn't too long winded!

Thanks for the help and time!
Dan
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Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

[Daniel Journo]

> Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp.
> AgentA answers and is able to use that feature code.
> If AgentA performs an attended transfer of a call from a queue to AgentB, the
> feature code no longer works.
>
> It only doesn't work when using Queue() and an Attended transfer is
> performed.
>
> Is this a bug or is there something that needs to be set to allow the
> DYNAMIC_FEATURES to be inherited after an attended transfer from a queue?

Doing some more tests, this reads like a bug to me.
Using a hanguphandler with DumpChan in the dialplan context that executes the 
Queue, I can see that DYNAMIC_FEATURES is set.
After the attended transfer when the call is ended, the hanguphandler still 
shows that DYNAMIC_FEATURES is set. It's just not accessible.

Any thoughts?
Thanks
Dan
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[asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

[Daniel Journo]

Hi,

I think I've identified an issue and just want to check before completing a bug 
report.

Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. AgentA 
answers and is able to use that feature code.
If AgentA performs an attended transfer of a call from a queue to AgentB, the 
feature code no longer works.

Cases that do work are as follows...
Calls using both Queue() and Dial() applications, prior to transfer, feature 
code works.
Calls using the Dial() application, both Blind and Attended transfers still 
allow the feature code to work.
Calls using the Queue() application where AgentA performs a Blind transfer to 
AgentB still allow the feature to work.

It only doesn't work when using Queue() and an Attended transfer is performed.

Is this a bug or is there something that needs to be set to allow the 
DYNAMIC_FEATURES to be inherited after an attended transfer from a queue?

Many thanks,
Dan

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Re: [asterisk-users] PJSIP redirect_method=uri_core and header modifications

[Daniel Tryba]

On Fri, Aug 03, 2018 at 04:24:06PM +0200, Daniel Tryba wrote:
> redirect_method=uri_pjsip works as expected with regard to the header
> manipulation stuff.
> 
> Also I can't remember why, in the past, I decided to not use uri_pjsip
> other than having the redirected host in CDRs with uri_core and the
> original with uri_pjsip (which I don't really care about).

I found that out after going live with uri_pjsip:
the domain in the to header with uri_pjsip is the hostname/ip of the
redirector. With uri_core the domain will be the hostname/ip of the
contact header in the 302 response.

Reading the redirect_method description (version 13):
"If this option is set to uri_core the target URI is returned to the
dialing application which dials it using the PJSIP channel driver and
endpoint originally used. If this option is set to uri_pjsip the
redirect occurs within chan_pjsip itself and is not exposed to the core
at all."

The wording for uri_core is that the original endpoint context will be
used, that would make changing the to domain a bug IMHO. But since I
need this behavior it is a nice feature.



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Re: [asterisk-users] PJSIP redirect_method=uri_core and header modifications

[Daniel Tryba]

On Thu, Aug 02, 2018 at 05:29:23PM +0200, Daniel Tryba wrote:
> With chan_sip there is the variable SIP_MAX_FORWARDS to set
> Max-Forwards. This counter is persistant after a redirect. I can't find
> the equivalent for PJSIP, so I went the way of header manipulation. Only
> to find out that any headers added to the outbound leg are lost after a
> redirect (with redirect_method=uri_core (didn't try any other since in
> the past they didn't work for me)).

redirect_method=uri_pjsip works as expected with regard to the header
manipulation stuff.

Also I can't remember why, in the past, I decided to not use uri_pjsip
other than having the redirected host in CDRs with uri_core and the
original with uri_pjsip (which I don't really care about).

I still think header manipulation should work with uri_core though.


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Re: [asterisk-users] 400 reply to INVITE not properly treated

[Daniel Tryba]

On Thu, Aug 02, 2018 at 02:40:48PM +1000, Patrick Wakano wrote:
> In my opinion, Asterisk should at fail the Dial and proceed with whatever
> was configured in the dialplan I tried some other 4XX SIP codes, but
> the only one I found not behaving properly is the 400 one

I think you are right, any 4xx is a final response. The call has failed at
this moment (unless there are other branches). You should file a bug report
IMHO.

https://tools.ietf.org/html/rfc3261#section-13.2.2.3
   A single non-2xx final response may be received for the INVITE.  4xx,
   5xx and 6xx responses may contain a Contact header field value
   indicating the location where additional information about the error
   can be found.  Subsequent final responses (which would only arrive
   under error conditions) MUST be ignored.

   All early dialogs are considered terminated upon reception of the
   non-2xx final response.

   After having received the non-2xx final response the UAC core
   considers the INVITE transaction completed.  The INVITE client
   transaction handles the generation of ACKs for the response (see
   Section 17).


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Re: [asterisk-users] PJSIP redirect_method=uri_core and header modifications

[Daniel Tryba]

On Thu, Aug 02, 2018 at 05:29:23PM +0200, Daniel Tryba wrote:
> With chan_sip there is the variable SIP_MAX_FORWARDS to set
> Max-Forwards. This counter is persistant after a redirect. I can't find
> the equivalent for PJSIP, so I went the way of header manipulation. Only
> to find out that any headers added to the outbound leg are lost after a
> redirect (with redirect_method=uri_core (didn't try any other since in
> the past they didn't work for me)).
> 
> Am I missing something? Or is this a PJSIP feature?

Forgot to mention this is tested on Debian/stable its Asterisk:
13.14.1~dfsg-2+deb9u3 (so 13.14.x with some fixes).


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[asterisk-users] PJSIP redirect_method=uri_core and header modifications

[Daniel Tryba]

With chan_sip there is the variable SIP_MAX_FORWARDS to set
Max-Forwards. This counter is persistant after a redirect. I can't find
the equivalent for PJSIP, so I went the way of header manipulation. Only
to find out that any headers added to the outbound leg are lost after a
redirect (with redirect_method=uri_core (didn't try any other since in
the past they didn't work for me)).

Am I missing something? Or is this a PJSIP feature?

chan_sip example:

[macro-maxforwards]
exten => s,1,NoOp()
exten => s,n,Set(mf=${SIP_HEADER(Max-Forwards)})
exten => s,n,Set(mf=$[ ${mf} - 1 ])
exten => s,n,ExecIf($[ ${mf} < 1 ]?Hangup(19))
exten => s,n,Set(__SIP_MAX_FORWARDS=${mf})
exten => s,n,SipAddHeader(X-Foo: bar)
exten => s,n,MacroExit()

[route]
...
exten => _+.,n,Macro(maxforwards)
exten => _+.,n,Dial(SIP/${EXTEN}@redirector)

An incomig INVITE will look like:

> INVITE sip:+number@asterisk;user=phone SIP/2.0
> Max-Forwards: 70

To the redirector:

> INVITE sip:+number@asterisk;user=phone SIP/2.0
> Max-Forwards: 69
> X-Foo: bar

< SIP/2.0 302 Redirect
< Contact: 

To somewhereelse:

> INVITE sip:+number@somewhereelse;user=phone SIP/2.0
> Max-Forwards: 69
> X-Foo: bar


PJSIP example (where the add strangely overrides the default
Max-Forwards: 70).

[setoutgoinglegvars]
exten => add,1,Set(PJSIP_HEADER(add,Max-Forwards)=60)
exten => add,1,Set(PJSIP_HEADER(add,X-Foo)=bar)
exten => add,n,Return()

[route]
...
exten => _+.,n,Macro(maxforwards)
exten => _+.,n,Dial(PJSIP/${EXTEN}@redirector,,b(setoutgoinglegvars,add,1))

> INVITE sip:+number@asterisk;user=phone SIP/2.0
> Max-Forwards: 70

To the redirector:

> INVITE sip:+number@asterisk;user=phone SIP/2.0
> Max-Forwards: 60
> X-Foo: bar

< SIP/2.0 302 Redirect
< Contact: 

To somewhereelse:

> INVITE sip:+number@somewhereelse;user=phone SIP/2.0
> Max-Forwards: 70



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Re: [asterisk-users] Do you set chan_sip's ignoresdpversion to true ?

[Daniel Tryba]

On Tue, Jun 19, 2018 at 07:38:12PM +0200, Olivier wrote:
> I've just discovered chan_sip's ignoresdpversion setting.
> Do you use it ?
> If positive which kinnd of issue could you solve with it ?

IIRC I used to enable this option when talking to some Ericsson SBC.  It
solved a problem concerning an on-hold scenario where the sess-version
wasn't getting incremented (by the Ericsson) even though SDP changed.
Asterisk would igore the re-INVITE without ignoresdpversion.

Today I use PJSIP with that SBC and I totally forgot this setting till
now, either the Ericsson was fixed/replaced or PJSIP handles the
scenario differently.


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Re: [asterisk-users] How to ignore REFER entirely with chan_sip or PJSIP ?

[Daniel Tryba]

On Fri, Jun 15, 2018 at 05:32:30PM +0200, Olivier wrote:
> In my testing, I saw that Asterisk always included a REFER value in each
> INVITE's Allow header, no matter how allowtransfer/allow_tranfer was set.
> 
> Is there a way to remove this REFER value entirely either globally or
> specifically for a given peer/endpoint ?

No, not with asterisk itself.

> Which telephony feature would loose without REFER method ?

AFAIK none except the ability to REFER (obviously).

What you can do is:
-route calls via a proxy that gives you the ability to modify SIP on
the fly (e.g. kamailio)
-wihh chan_sip use disallowed_methods (= REFER)
-with pjsip reject a REFER in the dialplan:
 https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Remote+Attended+Transfers

The last 2 options will make PBXs/endpoints that use REFERs, since
asterisk is advertising it, fail in certain scenarios. So cleanest is
the first option, which will take some work redesigning your setup but
might be a good thing on the long run.



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Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[Daniel Tryba]

On Tue, Jun 05, 2018 at 11:34:51AM +0200, Olivier wrote:
> 1.According SIP RFCs, is possible/recommended to have different values in
> From and P-Asserted-Id fields ?
> For instance, From field showing 123456789 and P-Asserted-Id showing
> 987654321 (beside privacy considerations) ?

Yes, most obviuos need for PAI is a call where anonimity is desired by
caller. Set the From to anonymous@anonymous.invalid and PAI to a real
user if the destination is trusted, any proxy that handles this message
that doesn't trust a destination will strip PAI thus ensuring privacy.
 
> 2. When Bob forwards to Cory a call coming from Alice, would expect
> Diversion/History-Info header to include Alice's number ?

No, diversion/history should contain Bob.

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Re: [asterisk-users] Long extensions that contain dashes

[Daniel Tryba]

On Tue, May 29, 2018 at 08:32:39PM -0700, David P wrote:
> We would like to use 20-char extension values that use dashes and alphanums
> after the first four digits. In order to handle these via pattern-matching,
> how can I define a pattern that allows dashes? There seems to be no option
> at http://the-asterisk-book.com/1.6/einleitung-regex.html#re
> gular-expression-syntax However, when I try a period, it seems to match the
> long suffix including the dashes. I want to know whether to depend on this
> continuing to work.

You should read some more up to date documentation I guess.
https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching
mentions the significance of - in a character class and the way to
escape. I'm to lazy to try this myself, but in any regexp parser I ever
used a litteral - in a charclass is defined by simply putting it as the
last character in the class (but above url leads me to think this
doesn't work in asterisk): 

[0-9] matches 0,1,2,3,4,5,6,7,8 and 9
[09-] matchers 0,9 and -
 
> Also, we're not sure whether our automated members can handle extensions
> longer than 4 digits. I'd like to pass a substring of our
> extension/destination_number in the call to Queue(). I couldn't find
> documention of any Queue() option like this. Is it possible to control the
> extension that the member receives?

You are looking to modify the callerid? e.g.:
Set(CALLERID(num)=${CALLERID(num):4})

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Re: [asterisk-users] Decoding SIP register hack

[Daniel Tryba]

On Thu, May 17, 2018 at 12:27:17PM -0400, sean darcy wrote:
> > WARNING.* .*: fail2ban=''
> >
> ># Option:  ignoreregex
> ># Notes.:  regex to ignore. If this regex matches, the line is ignored.
> ># Values:  TEXT
> >#
> >ignoreregex =
> >
> >
> Thanks. Very useful as a tutorial for fail2ban.
> 
> But I don't think it covers this SIP hack. This guy isn't trying to
> register.

His filter doesn't only trigger on REGISTERs, see the last line of the
matches and the context for guests (which logs the pattern of the last
line of the filter on an INVITE).

>  That why I find it puzzling. What is he trying to do ?

There are sip servers publicly reachable that will relay INVITEs, make
sure yours aren't. And there are only 2 kinds of operators of sip
server:
-those that have been the victim of toll fraud
-those that will be the victim of toll fraud

You can do nothing to stop this kind of traffic. The only thing you can
do is block it, either using only a whitelist (cumbersome) or generate a
blacklist with for example fail2ban or a more elaborate honeypot setup.
Or setup a proxy that will filter patterns you discover from 

BTW this is not a person, this is an automated script, running most
likely on compromised machines and sending spoofed ips. These scripts
care about generating a ring on a phone (again most an abuseable/hacked
account (or purchased with CC fraud)). If they find a server that does,
it will be targetted for all kind of fraud.

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Re: [asterisk-users] When should a Progress or Ringing be used in a today's telephony ?

[Daniel Tryba]

On Wed, May 16, 2018 at 04:51:49PM +0200, Olivier wrote:
> 1. When Asterisk receives a SIP call coming from PSTN, is there a time
> frame within which Asterisk must reply something to keep caller from
> canceling the call ? Where does this limit come from ? From SIP RFC ? From
> local regulation bodies ?
> 
> 2. Which SIP signal is required to stop call cancellation in the previous
> case ?

See RFC 3261, 17.1.1. A (provisional) response to an INVITE is required
within a timelimit. After a provisional response a non-provisional
response is required. Defaults are on page 264 of the RFC (first to
last).
 
> 3. When Asterisk receives a call, either from PSTN or from a SIP phone) it
> cannot process (unkown callee, whatever reason, ...), should you stop
> processing with Hangup or Congestion ?
> Hangup application allow for exit code customization, if I'm not mistaken,
> but  Congestion exists for a reason.

With regard to PSTN calls the signalig is limited, but to a SIP device
you could signal usefull information, eg: unknown, temp. unavailable.
Why not give a usefull reason instead of Congestion

> 4. Is it a good practise to send a 180/183 when you don't get one ?

People will complain if there is no indication, so yes IMHO.

> 5. I observed I sometimes got  a 100 Trying then a 183 session Progress
> when outcalling some (mobile) phones while simpy getting 100 Trying when
> some other (mobile) phone through the same carrier (most probably, end
> devices were not managed by the same (mobile) telephony provider).
> What explains such difference ?

An explanation could be packet loss. But there are no requirements for
1xx responses to an INVITE. Maybe they just don't care about feedback to
callers.


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Re: [asterisk-users] Streaming MoH from iHeart radio?

[Daniel Tryba]

On Wed, May 16, 2018 at 11:01:53AM -0400, Mike Diehl wrote:
> I have a user who would like to stream their favorite radio station from
> iHeart radio for their music on hold.
> 
> It this TECHNICALLY possible?

Yes.

> If so, any pointers would be appreciated.

https://www.voip-info.org/asterisk-config-musiconholdconf/#Exampleusingicecastampshoutcaststreams
 

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Re: [asterisk-users] SIP Codec negotiation

[Daniel Tryba]

On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote:
> I receive an INVITE/SDP containing:
> 
>   m=audio 11310 RTP/AVP 3 0 101
> 
> which I interpret as gsm, ulaw, rfc2833.
> 
> and I reply with an OK/SDP containing:
> 
>   m=audio 15884 RTP/AVP 0 3 101
> 
> which I interpret as ulaw, gsm, rfc2833.
> 
> How can I tell which codec was actually used for the call?

AFAIK this is undetermined. The callee can send either ulaw or gsm,
unless the caller wants to narrow it down to 1 codec, see
https://tools.ietf.org/html/rfc4317#section-2.2

Most of the time the callee will pick the first (so in this case ulaw).
But there are media gateways out there that choose g711[au] above "more
complex" codecs regardless order in SDP. My prefer PSTN provider will
always prefer alaw if offered since that will prevent transcoding on
their side if the call goes to ISDN/POTS, but AMR if the call goes to
VoLTE.


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Re: [asterisk-users] multi step auth?

[Daniel Tryba]

On Tue, May 08, 2018 at 03:04:55PM -0500, Jeff LaCoursiere wrote:
> Thats till doesn't change the SIP header.?? Basically they want to send a RE
> INVITE and authenticate my DID number.?? But my DID number does not have a
> peer or user entry in sip.conf.?? Perhaps I am answering my own question,
> but is that the only way this is going to work?

Maybe you should post their requirments (instead of your rephrasing of
them). Do they actually want to have different from/to and contact(!) in
one SIP dialog? But AFAIK you don't have such control in Asterisk, you
can only influence the original INVITE and than have Asterisk respond
to a auth challenge, which you can influence with defaultuser according
to sip.conf.

So experiment with something like
[user]
fromuser=thenumber
defaultuser=theusername
remotesecret=thepassword

and see what the fromuser in request is and what the authentication user
in the Authorization header is in step 3, according to sip.conf remarks
it should be:

From: 
To:
Authorization: Digest username="defaultuser"


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Re: [asterisk-users] Pass through registration / proxy

[Daniel Tryba]

On Wed, Apr 11, 2018 at 12:04:18PM -0400, Telium Technical Support wrote:
> Maybe proxy is the wrong word I chose.  Asterisk is something like a peer to
> the legacy PBX.  I thought about setting up individual SIP accounts on the
> Asterisk box to connect to the legacy PBX, or maybe a SIP trunk to the
> legacy PBX (assuming it can route calls through the SIP trunk to a peer to
> reach a phone).  The legacy PBX is a Nortel in case that matters.
> 
> I'm supposed to figure this out and present options but having trouble
> figuring out if Asterisk would be a peer, or pretend to be many sip agents
> registering on the legacy Sip pbx, etc.  I think I'm stuck at the conceptual
> level.  (Still a beginner in training - but having fun learning Asterisk)

One of my first integrations was similar but with a Siemens. Easiest
might be a SIP trunk (peer) between Asterisk/Nortel and have different
prefixes for the Norted (1xx) and Asterisk (2xx) and route these to the
other. The SIP endpoints simply register to Asterisk.


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Re: [asterisk-users] Pass through registration / proxy

[Daniel Tryba]

On Tue, Apr 10, 2018 at 09:22:02PM -0400, Telium Technical Support wrote:
> I need to create a SIP proxy to be placed in front of a legacy PBX.  When a
> phone registers with the proxy, I would like Asterisk to register with the
> PBX behind it.  (To tell the PBX to send calls to the proxy and then to the
> SIP phone).
> 
> Can I use Asterisk to create a proxy like this?  Is there a way to cause the
> Asterisk to register with another host when it receives a successfully
> registration?

You can, but maybe you should use a sip proxy (like kamailio) for this
task instead of a back to back user agent like asterisk.

You can listen to events triggered on registration to asterisk and with
realtime intergration add the register to the PBX (or manipulate
sip.conf). This still might be easier to implement compared to (for
example) kamailio if you are new to that.

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Re: [asterisk-users] PJSip CallerID Question

[Daniel Tryba]

On Fri, Apr 06, 2018 at 02:27:31PM -0500, Brent Davidson wrote:
> I have multiple Asterisk instances set up in different locations and would
> like to modify the callerID of inbound calls to identify which instance the
> call is coming from.  I knew how to do that with the old sip format, but
> can't seem to figure it out with PJSip.

So how did you do that?
 
> Currently Location A, extension 10 calls Location B, extension 20.  CallerID
> on Extension 20 displays "10" for the callerID.
> 
> The Desired configuration is for Extension 20 to show "Locati0n B - 10" on
> the caller ID.  I don't want to modify the caller ID for each individual
> extension as I want the intra-location caller IDs to show just the extension
> number.  (e.g. LocA/Ext. 10 calls LocA/Ext11 - LocA/Ext11's CallerID
> displays "10", but LocA/Ext10 calling LocB/Ext20 displays "Location A - 10"
> for caller ID.

You examples contradict.
 
> Rather than routing these to the "internal" context, should I create another
> context and somehow parse/manipulate the caller ID in there then route to
> "internal" ?

TIMTOWTDI, but I'd choose to set the CALLERID(name) on the sending side
dialplan (where it routes calls to external extensions).

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Re: [asterisk-users] Setting outgoing CALLERID without changing CDR(src)

[Daniel Tryba]

On Wed, Mar 28, 2018 at 08:16:26PM -0600, Carlos Chavez wrote:
> ?? I thought I had found and answer to this question by using
> CALLERID(ani) but it seems that only works on versions prior to 12.?? On
> Asterisk 13 setting CALLERID(num) before dialing to an external trunk always
> changes CDR(src) to the number you set and the original extension number
> that dialed is lost.?? How are you handling this??? Am I forced to use a
> custom field to keep the original caller number??? My billing software uses
> the src field to get the extension that dialed the call.?? Any tips?

What does your external trunk use as sources for callerID? For example
if your provider supports PAI and sets this headers as callerID, your
problem is solved. Otherwise you could for example use the custom csv
output and set a custom CDR field (that stores the original src) in the
place of src.

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Re: [asterisk-users] Is 100 trying mandatory? Can asterisk answer with 180 without prior 100 trying?

[Daniel Tryba]

On Mon, Mar 19, 2018 at 12:59:47PM -0300, Joshua Colp wrote:
> > To try to reproduce the problem with our SBC, is there a way to tell
> > the asterisk, preferably PJSIP, to directly answer with 180 ringing
> > without prior 100 trying?
> 
> The PJSIP channel driver has no option or ability to do this. I do not recall 
> if chan_sip does.

A (very) dirty workaround would be to drop these packets with iptables
(assuming Linux as OS), something like:

iptables -t raw -I OUTPUT -p udp -d ipaddrofpbx -m string --algo bm  --from 0 
--to 32 --string "SIP/2.0 100 " -j DROP

Don't try it with TCP :)


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Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

[Daniel Tryba]

On Wed, Mar 07, 2018 at 10:08:52PM +, Thomas Peters wrote:
> You did indeed warn me. I've made progress, gotten the dhcp option 242 to 
> work, and finally gotten the phone to the point where it asks for a username 
> and password. I defined these on the Asterisk server. I entered them on the 
> phone. It says "Acquiring Service" and sits there.  At least it sets the 
> clock on the phone now. 
> 
> TCPDump shows traffic going back & forth every few seconds: 
> 10.1.138.245.13602 > ad-apbx.mcts.org.sip
> ad-apbx.mcts.org.sip > 10.1.138.245.60206
> 10.1.138.245.30360 > ad-apbx.mcts.org.sip:
> 
> But it sits at Acquiring Service eternally. 
> 
> The username and password I am entering are the Asterisk extension and 
> asterisk extension secret. Should these be entered somewhere in the craft 
> menu instead?

My 9608 has no password/secret in asterisk, could be due to some
problem like you are encountering. Also these phones only work with SIP
over TCP.

sip.conf for my 9608:

[a204]
type=friend
context=204
defaultuser=204
host=dynamic
qualify=yes
nat=yes
disallow=all
allow=alaw
dtmfmode=rfc2833
canreinvite=no
transport=tcp,udp


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Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

[Daniel Tryba]

On Tue, Mar 06, 2018 at 05:36:04PM +, Thomas Peters wrote:
> But please don't tell me the only way to program up each phone is via
> the craft interface?
> 
> Every other phone I've ever used requires a configuration file, which
> has the MAC address of the phone as its name. The Avaya phones must
> have some other method. Unless I have to embed the mac address and
> particulars for all the phones into the 46xxsettings.txt file??

I tried to warn you, didn't I? :)

The phones themselves are nice, when used with an Avaya PBX. What I have
seen is that these phones are really dumb themselves and need a decent
PBX that does the smartstuff via proprietary interfaces (H.232
gatekeeper like IPOffice or whatever they use with the SIP based
Communication Manager)

There is some intelligence in parsing the 46xxsettings file, but AFAIK
you need the CRAFT menu to do this (apart from MAC adresses).

My advise: sell these phones, any SIP device you can buy with the
proceeds is more intelligent when used with Asterisk.

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Re: [asterisk-users] Half Off Topic Questions

[Daniel Tryba]

On Tue, Mar 06, 2018 at 09:05:25AM +0100, Markus Weiler wrote:
> we're just wondering, in German we call the different types of phone-numbers 
> (Geographic,mobile,national,VoIP...)
> Rufnummerngassen (phone number alleys ;-) )  
> Is there an english word for this? 

I'd call it something like "breakout type", but a more literal
"extension (type) path" might also convey the idea.

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Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

[Daniel Tryba]

On Thu, Mar 01, 2018 at 02:46:31PM +, Thomas Peters wrote:
> Right-- I've seen the Avaya document you cite below. It says "To
> administer DHCP option 242, make a copy of an existing option 176" but
> I don't have any example of option 176 or 242 to copy, and don't know
> what to do to /etc/dhcpd.conf to make it offer option 242. 
> 
> Then there's this long table of parameters to use with (presumably)
> option 242.
> 
> I was hoping someone had a working minimal example of a dhcp option
> 242 config I could copy and modify.

Example for our old IP Office (192.168.250.1)  setup:

option option-242 code 242 = string;

subnet 10.0.0.0 netmask 255.255.0.0 {
   #option option-242 "L2Q=1,L2QVLAN=4";
   #option option-242 "HTTPSRVR=10.1.2.3/files";
   option option-242 "MCIPADD=192.168.250.1,MCPORT=1719,HTTPSRVR=192.168.250.1";
}


My guess is you only need HTTPSRVR=hostname/path 
This should point to a dir where a 46xxsettings.txt exists.
Must contain something like:


SET SIPPROXYSRVR 172.16.0.2
SET SIPPORT  5060
SET SIPDOMAIN 172.16.0.2
SET SIPREGISTRAR 172.16.0.2
SET SIP_CONTROLLER_LIST 172.16.0.2:5060;transport=tcp 
SET ENABLE_AVAYA_ENVIRONMENT 0 
SET CONFIG_SERVER_SECURE_MODE 0
SET SIPSIGNAL 0
SET REGISTERWAIT 900

SET CLDISPCONTENT 0
SET DISPLAY_NAME_NUMBER 3

SET DIALPLAN 2xx|0[1-7]|08[458]xxx
SET PHNNUMOFSA 2
SET GMTOFFSET 1:00
SET DSTOFFSET 1
SET DSTSTART LSunMar2L
SET DSTSTOP LSunOct2L

SET BRANDING_VOLUME 1
SET AGCHAND 0
SET AGCHEAD 0
SET AGCSPKR 0
SET DATETIMEFORMAT 2
SET TIMEFORMAT 1
SET SNTPSRVR 109.235.32.103,109.235.32.119
SET ENTRYNAME 0

SET PHNOL


 
> They only have minimal function? No speed dials, BLFs, etc? 

Not as fas as I know. I configured this in 2013 and at that time only
call/transfer and conference worked.


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Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

[Daniel Tryba]

On Wed, Feb 28, 2018 at 08:48:38PM +, Thomas Peters wrote:
> I'd like to start configuring my Avaya 9608G phones for use on
> Asterisk / FreePBX / PBX-In-a-Flash. I'm using a variety of other
> phones on my system without major issues.
> 
> I've read the discussion back in March, May and August of 2016, but
> unfortunately, my difficulty is much more basic. I think it has to do
> with DHCP, specifically, what options I'm offering the phone via DHCP.

So you might want to start without configurations from DHCP. Enter the
file/http server in the phone manually and point it to a http server
containing the needed firmware files and a correct 46xxsettings.txt for
you asterisk.

Avaya uses other dhcp options for these phone (242):
https://downloads.avaya.com/elmodocs2/one-X_Deskphone_Edition/R1.5/output/16_300698_4/admn054.html

BTW these phones are a terrible waste when used with Asterisk. They only
provided minimal functionality (calls and transfer).

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Re: [asterisk-users] asterisk mysql contacts

[Daniel Tryba]

On Wed, Jan 17, 2018 at 03:16:04PM +0200, Atux Atux wrote:
[asterisk dialplan mysql]
> I would like to ask if there is a way to implement this easily in my
> dialplan, please.

The answer is: yes

If you'd search for "asterisk dialplan mysql", you'get something like
https://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL

You might want to look into AGI plus your favorite scriptinglanguage
though.

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Re: [asterisk-users] remote Asterisk console

[Daniel Tryba]

On Tue, Jan 16, 2018 at 06:19:30PM +0100, Paul Neuwirth wrote:
> Thank you both. That was (most likely) what I was looking for - but
> still some worries about sending plaintext passwords...

The AMI interface can use a Challenge-Response mechanisme for logins,
if you are this concerned you should use this even over TLS/SSL/SSH.


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Re: [asterisk-users] Digium G100 and CID Dropping First Digit.

[Daniel Tryba]

On Mon, Jan 15, 2018 at 10:41:27PM +, David Klaverstyn wrote:
> port1 < Calling Number (len=12) [ Ext: 0  TON: National Number (2)  NPI: 
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> port1 <   Presentation: Presentation allowed of 
> network provided number (3)  '21xx' ]
> port1 < [70 0a c1 30 34 39 31 34 31 32 31 34]
> port1 < Called Number (len=12) [ Ext: 1  TON: Subscriber Number (4)  NPI: 
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '049xx' ]
> 
> -- Accepting call from '21xx' to '049xx' on channel 0/5, span 1

Don't know anything about the card you are using, but seeing ISDN
signaling that the type of number (TON) is national and that means
overhere that leading zeros are stripped, I see nothing wrong with it.
Looking at my old chan_dadhi configs there are options to prefix
something based on TON. So over here I have configured:
nationalprefix = 0
to prefix the leading 0 for national numbers that callees expect.
The G100 manual contains the phrase "national prefix", but no info about
it, so look into those prefix options.

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Re: [asterisk-users] Can't install package asterisk-dbgsym on Stretch

[Daniel Tryba]

On Fri, Dec 08, 2017 at 06:11:47PM +0100, Olivier wrote:
> 1. Is this a bug in debian-debug repo ? If positive, should I file a bug
> report ?
> 
> 2. Is correct to understand that to get DONT_OPTIMZE, BETTER_BACKTRACE and
> so on options compiled in, I must recompile anyway ?

As far as I know the deb debug package just contains debugging symbols
(before these are stripped), you still need the normal asterisk packages
with the binaries. The last time I tried to debug with these packages it
was a waste of time since your not running a version with debugging
options. These debug packages show some extra info in gdb traces, but
lots of important stuff is still optimized away. 

If you need to debug install the source package and compile from there.
You might want to figure out how to manipulate the deb build rules to
include wanted flags, shouldn't be to hard but it has been a few years
for myself.


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[asterisk-users] PJSIP_AOR Slow

[Daniel Journo]

Hi,

In my dialplans, I'm currently using PJSIP_AOR to check the status of a contact 
before dialling so that I can route the call differently if the endpoint is 
offline.
But PJSIP_AOR seems to take about 0.9 seconds to return. If I'm checking 10 
endpoints, that can cause a significant delay.

Is there a better way to check the status of an endpoint pre-dialling within 
the dialplan?

Here is a sample of what I'm doing.

exten => 
example_839,9,ExecIf($["${PJSIP_AOR(example_220,contact)}"=""]?Set(UNAVAILABLEPEER=${UNAVAILABLEPEER}
 example_220))
exten => 
example_839,10,ExecIf($["${PJSIP_AOR(example_220,contact)}"!=""]?Set(WORKINGPEERFOUND=1))
exten => example_839,11,NoOp(${PJSIP_AOR(example_223,contact)})
exten => 
example_839,12,ExecIf($["${PJSIP_AOR(example_223,contact)}"=""]?Set(UNAVAILABLEPEER=${UNAVAILABLEPEER}
 example_223))
exten => 
example_839,13,ExecIf($["${PJSIP_AOR(example_223,contact)}"!=""]?Set(WORKINGPEERFOUND=1))
exten => example_839,14,NoOp(${PJSIP_AOR(example_224,contact)})
exten => 
example_839,15,ExecIf($["${PJSIP_AOR(example_224,contact)}"=""]?Set(UNAVAILABLEPEER=${UNAVAILABLEPEER}
 example_224))
exten => 
example_839,16,ExecIf($["${PJSIP_AOR(example_224,contact)}"!=""]?Set(WORKINGPEERFOUND=1))
exten => example_839,17,NoOp(${PJSIP_AOR(example_226,contact)})
exten => 
example_839,18,ExecIf($["${PJSIP_AOR(example_226,contact)}"=""]?Set(UNAVAILABLEPEER=${UNAVAILABLEPEER}
 example_226))
exten => 
example_839,19,ExecIf($["${PJSIP_AOR(example_226,contact)}"!=""]?Set(WORKINGPEERFOUND=1))
exten => example_839,20,NoOp(${PJSIP_AOR(example_227,contact)})
exten => 
example_839,21,ExecIf($["${PJSIP_AOR(example_227,contact)}"=""]?Set(UNAVAILABLEPEER=${UNAVAILABLEPEER}
 example_227))
exten => 
example_839,22,ExecIf($["${PJSIP_AOR(example_227,contact)}"!=""]?Set(WORKINGPEERFOUND=1))
exten => example_839,23,NoOp(${PJSIP_AOR(example_240,contact)})
exten => 
example_839,24,ExecIf($["${PJSIP_AOR(example_240,contact)}"=""]?Set(UNAVAILABLEPEER=${UNAVAILABLEPEER}
 example_240))
exten => 
example_839,25,ExecIf($["${PJSIP_AOR(example_240,contact)}"!=""]?Set(WORKINGPEERFOUND=1))
exten => example_839,26,GotoIf($[${WORKINGPEERFOUND}=0]?227)

Many thanks
Dan

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Re: [asterisk-users] Gerrit usage?

[Daniel Tryba]

On Fri, Sep 29, 2017 at 12:27:53PM -0300, Joshua Colp wrote:
> > "git checkout -b 13" appears to fix this.
> 
> This did not create a branch from 13. This created a branch named "13"
> from the branch you were on, which was most likely master. That is why
> your "git review" is not working as you expect, because you are telling
> it that you did the work against "13" but it really was against master.
> 
> git checkout -b 13 origin/13
> 
> Would create a local branch "13" which is from the remote branch "13".
> You'll need to do this, or do your "git review" against master and then
> cherry pick from inside Gerrit to the appropriate branches.

Thank you for your near instant feedback, this fixed my problem and I
was able to submit code for a review.



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[asterisk-users] Gerrit usage?

[Daniel Tryba]

I'm trying to figure out how to commit some code for review. Following:
https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage

Created a ssh alias.
Cloned using: "git clone ssh://asterisk/asterisk"
Set name and email.
Installed the gerrit commit hook: "git review -s"
Try to change to asterisk 13 for creating a patch: "git checkout 13"
This fails with:
error: pathspec '13' did not match any file(s) known to git.


"git checkout -b 13" appears to fix this.

Created a new branch:
git checkout -b ASTERISK-27284

Did some work, added and commited this work. So far so good.

Now trying to submit this: "git review 13"
Fails with:

>Errors running git rebase -p -i remotes/gerrit/13
>error: could not apply 5760526... Update UPGRADE.txt for 13 branch
>
>When you have resolved this problem, run "git rebase --continue".
>If you prefer to skip this patch, run "git rebase --skip" instead.
>To check out the original branch and stop rebasing, run "git rebase
>--abort".
>Could not pick 5760526f69ad02189c8e385e2e974be4cba11b6e
>It is likely that your change has a merge conflict. You may resolve it
>in the working tree now as described above and then run 'git review'
>again, or if you do not want to resolve it yet (note that the change can
>not merge until the conflict is resolved) you may run 'git rebase
>--abort' then 'git review -R' to upload the change without rebasing.


Somehow some way, UPGRADE.txt and UPGRADE-13.txt are changed and I can't
find any way to discard/ignore/remove/skip these changes. Clearly I
don't understand git and the way it handles conflicts it created by
itself.

What is going wrong? What is the magical git command to just commit the
2 files I added/commited for review?

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

[Daniel Tryba]

On Thu, Aug 31, 2017 at 05:54:43PM +, Joseph Smith wrote:
> 
> So I am looking for a better way to allow several thousand callers to listen 
> to this IVR menu at the same time.
> 

An alternative that comes to mind is to have 1 conference with 1 channel
playing MoH in it and then add callers in a muted state to it. Never
tried this, don't know if it fits your case.


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Re: [asterisk-users] Pass CallerId/Privacy info from A Leg to B Leg

[Daniel Tryba]

On Thu, Aug 17, 2017 at 07:28:00AM +, Grant Bagdasarian wrote:
> Is there an option to give to the Dial command, or another variable to set, 
> to make Asterisk copy such information to the B Leg?
> Or do I have to program this out myself?

In chan_sip there are the trustrpid and sendrpid option:

;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = pai ; Use the "P-Asserted-Identity" header
; to send the identity of the remote party 

In pjsip:
;trust_id_inbound=no; Accept identification information received from this
; endpoint (default: "no")
;trust_id_outbound=no   ; Send private identification details to the endpoint
; (default: "no")
;send_pai=no; Send the P Asserted Identity header (default: "no")


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Re: [asterisk-users] Change OS from CentOS 6 to 7

[Daniel Tryba]

On Fri, Aug 04, 2017 at 03:27:40PM -0400, Jerry Geis wrote:
> Audio packets are running...
> 
> 961 16.150421076 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU,
> SSRC=0x6A3E0AF1, Seq=28402, Time=73280
> 962 16.170411284 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU,
> SSRC=0x6A3E0AF1, Seq=28403, Time=73440
> 963 16.190381989 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU,
> SSRC=0x6A3E0AF1, Seq=28404, Time=73600
...

Which is only one way!

Your info is lacking any useful information, that makes helping you
extremly difficult. Enable debugging in asterisk (core set verbose 3,
sip set debug on and rtp set debug on) and compare what asterisk is
seeing to packet captures. Tell us what ip adress is what device, try
telling us what you are trying to accomplish (eg are you calling an echo
test).

Have you tried turning of selinux? If that solves the issue, take a look
at how to actually set ip up correctly for asterisk.


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[asterisk-users] Moving call DAHDI from channel X to Y.

[Daniel Harper]

Thanks for the response. Investigating the logs I think the (outbound)
channel is being moved just after a inbound call.

We have a AMI application that sits on top of asterisk and we want to
know when a channel is moved. (i.e. we might move the agent channel
and outbound channel into a meetme room to do playback etc)

I guess the only option we have would be to patch the source to
generate a custom AMI event which would be difficult to test given the
it happens perhaps 1-2 times per 10,000-20,000 calls!

Thanks
Daniel


>This is a normal thing that can happen when doing B-channel selection
>as a part of the call setup process.  If one side disagrees with the
>initial choice for B-channel, the other side can request that it be
>moved in the reply.  It's more likely to happen on busy PRIs with
>bidirectional (ingress and egress) traffic.

>Hope that helps!

>--
>Matthew Fredrickson
>Digium, Inc. | Engineering Manager
>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-users] Moving call DAHDI from channel X to Y.

[Daniel Harper]

I am seeing the in the asterisk logs that channels (PRI ISDN)  are
being moved ..

[Jul 29 16:31:48] VERBOSE[16125] logger.c: -- Moving call
(DAHDI/57-1) from channel 57 to 58.

I then see the moved channels with a "0:" in front of it.

[Jul 29 16:31:48] VERBOSE[26691] logger.c: -- Hungup 'DAHDI/0:58-1'

Any ideas why this could be happening?

I believe these messages are coming from chan_dahdi.c and the
"pri_fixup_principle" function.

-- 
Cheers,

Daniel

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Re: [asterisk-users] Support for inbound UPDATE request

[Daniel Tryba]

On Fri, Jul 07, 2017 at 07:44:26PM +0530, Rahul MathuR wrote:
> Could you please let me know whether the latest Asterisk has a support for
> inbound UPDATE ?
> 
> In my case, the carrier is sending an UPDATE to change the codec which is
> replied by 5xx from Asterisk 11.17.1.

Asterisk 13/PJSIP supports inbound UPDATEs, I see them all the time as
inbound Session-Timer mechanism.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_direct_media_method
mentions support for SDP renegotiation via UPDATE if supported by the
endpoints being preferred above re-INVITEs.


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Re: [asterisk-users] PJSIP equivalent for SIPDtmfMode?

[Daniel Tryba]

> > Can't find a way to control the dtmf mode on a per session basis with
> > pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any
> > hints on how to do this?
> 
> There is no current way, but a community member has recently posted a
> change[1] for review which implements this.
> 
> [1] https://gerrit.asterisk.org/#/c/5909/

Just what I wished for. I'd love to see this added.

But I see later versions (>13.14.1) have an dtmfmode auto_info option,
might also fit my needs.

Thanks for your feedback to my question(s).


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Re: [asterisk-users] DMTF payload bug in 13.14.1 with pjsip and direct_media?

[Daniel Tryba]

On Thu, Jun 29, 2017 at 11:55:51AM -0500, Richard Mudgett wrote:
> > To me this looks like a bug in asterisk. Either asterisk should use the
> > same rtp payloads for telephone-events on both call legs during inital
> > callsetup or asterisk should come to the conclusion there is an
> > incompatible "codec" on both legs so it shouldn't switch to direct
> > media.
> >
> > Has anyone else seen this issue?
> >
> 
> This is an old issue.  One of the latest issues is:
> 
> https://issues.asterisk.org/jira/browse/ASTERISK-25166

I was looking DTMF related problems and found none. Looks like it is a
more general issue related to all capabilities. Thank you for pointing
this out. 

Seeing the history of the bugs the problem and the full fix is larger
than I initially thought. Maybe a quick stopgap is to just not try to
setup direct media if there are numeric differences between call legs
(this would help me since most call would be direct media, I'll try to
look into this is I have the time to look into this and find out if I
have enough knowlegde to try something).


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[asterisk-users] PJSIP equivalent for SIPDtmfMode?

[Daniel Tryba]

Can't find a way to control the dtmf mode on a per session basis with
pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any
hints on how to do this?


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[asterisk-users] DMTF payload bug in 13.14.1 with pjsip and direct_media?

[Daniel Tryba]

While trying to use direct_media I'm seeing RTP payload mismatches after
succesful reinvites.

Initial INVITE from endpoint A to asterisk has rfc4733 DMTF
m=audio 35648 RTP/AVP 9 8 111 96
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16

>From asterisk to upstream U:
m=audio 14338 RTP/AVP 9 8 111 18 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

So the payload types in the RTP streams from A and to U differ. This
works fine when asterisk is relaying media.

With direct_media=yes there are reinvites sent from asterisk to both A
and U. The invite to A contains:
c=IN IP4 ipaddrofU
m=audio 33142 RTP/AVP 8 96
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16

And the invite to U contains:
c=IN IP4 ipaddrofA
m=audio 35648 RTP/AVP 9 8 111 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

Both sides respond with a 200 OK and asterisk is not
relaying/transcoding the media anymore. At this moment DTMF send from A
isn't getting recognized by U, which IMHO is totally understandable
since U doesn't know about payload 96. 

To me this looks like a bug in asterisk. Either asterisk should use the
same rtp payloads for telephone-events on both call legs during inital
callsetup or asterisk should come to the conclusion there is an
incompatible "codec" on both legs so it shouldn't switch to direct
media.

Has anyone else seen this issue?

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Re: [asterisk-users] Writing CDR's to two database servers

[Daniel Tryba]

On Mon, Jun 19, 2017 at 11:47:04AM -0400, Tech Support wrote:
> 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.

Beside the already mentioned solutions, you could take a look at a
multi-master setup like a Galera Mysql cluster. Works perfectly for
short queries without locks like CDRs.


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Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

[Daniel Tryba]

On Fri, Jun 16, 2017 at 08:38:59AM +0100, J Montoya or A J Stiles wrote:
> > Whatever has been done, if anything, isn't working effectively.  At this
> > point I'd like to see some response from the mailing list admin about any
> > root-cause efforts, AFAIC this is starting to smear the Digium/Asterisk
> > brand's ability to handle IT related issues...  No response = no confidence
> > vote.
> 
> It's hardly Digium's fault,

Actually it is. They are pretending to send email from our/your/my
emailadresses without taking the proper steps how to do this in a modern
age.

[snip google rant]

DMARC reports inform me that most rejections come from Google (500+),
Microsoft has far far less rejection (less than 10 IIRC), then comcast
and some other mail providers. It is just that most people (choose) use
Google, get over it.

What Google (and many many others) is doing is for the benifit of reducing
email spoofing and spam. Proper SPF/DKIM/DMARC are a must if you want to
send mail to the big parties. The time you could simply run your own
smtpd without any cares are long since gone, you need to comply to
current SMTP related RFCs to get mail accepted.

I'm still maintaining the idea that simply enabling DKIM signing on this
list solves the problem. It is supported by the MTAs I can see in the
headers and I linked to a howto in the past.

But Digium doesn't need to have this kind of knowledge, their business
is not SMTP based but SIP based (and I think they are great at that
business). But since the mailinglists are supplemental support services
it would be in their best interest to fix this mess in some way.


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Re: [asterisk-users] Is this the future of telephony?

[Daniel Tryba]

On Thu, Jun 15, 2017 at 08:56:29PM -0400, Christopher van de Sande wrote:
> I just setup an anonymous endpoint in pjsip.conf and a context that
> forwards to $EXTEN and when I setup the correct SRV records, it seems
> that any SIP client that's smart enough can just dial my SIP/email
> address.  Is this what the future looks like?

Look forward to a lot of SPIT (SPAM over Internet Telephony).

BTW how to you expect to get the data on your mobile device? That is a
"traditional" telephone company its bussiness, providing network
capabilities to endusers.

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Re: [asterisk-users] Asterisk 1.6.2 how to debug T.38 udptl problems

[Daniel Tryba]

On Thu, Jun 15, 2017 at 12:11:36PM +0200, Benoit Panizzon wrote:
> Or does anyone have an idea over what the asterisk is stumbling?

What if you set the maxdata in asterisk to a value lower than the other
side? e.g. sip.conf: 
t38pt_udptl = yes,fec,maxdatagram=400

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Re: [asterisk-users] CallerId presence issue

[Daniel Tryba]

On Wed, Jun 14, 2017 at 10:18:19AM -0400, Mike wrote:
> I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP.
> PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has
> its own callerid values and presence.  I pass on those calls to PBX_B via
> SI, and I'm trying to pass on this CALLERID info to PBX_B as well. 
> 
> My relevant dialplan snippet on PBX_A is:
> exten =>
> 1,1,Dial(SIP/pbx_b/55,,f(${CALLERID(all)})u(${CALLERID(pres)})))
...
> I'm clearly missing something to pass on the callerid presence state via
> the SIP link, but I can't figure out what.

Never heard of this method, are you sure this works for SIP, sound more
like for ISDN (look at packet captures).

But the/a standardized method is to use the P-Asserted-Identity and
Privacy headers (rfc3325). This should work if you set in the peer configs 
in sip.conf on both sides:
sendrpid=pai
trustrpid=yes

Or you can do header manipulation/getting/setting manualy if desired.

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Re: [asterisk-users] German sip dial rules

[Daniel Tryba]

On Mon, Jun 12, 2017 at 05:00:31PM +0200, Hans-Peter Jansen wrote:
> is somebody attending, that wants to share his outgoing dial rules of 
> extension.conf, like used in typical(?) german pbx setups?
> 
>  * zero prefix for outside calls
>  * zero zero or plus prefix for international calls
>  * handle emergency calls
> 
> With ISDN, one was able to just forward the called number, but with sip, one 
> has to normalize the dialed pattern in order to match SIP (provider) 
> expectations... As always, the devil is in the details.

Shouldn't you just ask the provider?

But not being German, the only problem I know of is the ISDN 
sub addressing feature widely in use.

Looking at https://en.wikipedia.org/wiki/Telephone_numbers_in_Germany
I'd guess a dialplan would be (assuming the operator wants e164+):

exten => _1.,1,Dial(SIP/${EXTEN}@provider)
exten => _[2-9].,1,Goto(+49xyz${EXTEN:1},1)
exten => _0[1-9].,1,Goto(+49${EXTEN:1},1)
exten => _00[1-9].,1,Goto(+${EXTEN:2},1)

exten => _+.,1,Dial(SIP/${EXTEN}@provider)


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Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

[Daniel Tryba]

On Mon, Jun 12, 2017 at 09:07:31AM +0200, Olivier wrote:
> 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 ?

See my DMARC thread message:
http://lists.digium.com/pipermail/asterisk-users/2017-June/291545.html

Any DMARCed domain used for sending will cause bounces for recipients on
platforms that check and honor the DMARC configuration.

There may ofcourse be other causes.

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Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

[Daniel Tryba]

On Sun, Jun 11, 2017 at 01:16:10PM +0200, Michael Maier wrote:
> Let's go into details:
> I'm starting at the point where asterisk / fax client receives the T.38
> reininvite from the server from the provider 195.185.37.60:5060 for the
> fax client (extension 91):

I'm running Asterisk 11 on my faxserver so not using pjsip but chan_sip.
But IIRC I had problems with asterisk-11/t38modem-2.0.0/hylafax if the
upstream side started the t38 reINVITE on sending a fax. My hylafax
config.ttyT38* contains the AT F command to initiale t38 on the t38modem
side for outgoing calls. For incoming t38modem also starts the reINVITE
but there is no parameter to influence this in my configs.

No idea if this is in anyway related nor what the default is of the
options below and neither if it depends on the t38modem version.

# T.38 dial modifiers
#
# F - enable T.38 mode request after dialing
# V - disable T.38 mode request after dialing (remote host should do it)
#
# calling/called number dial modifiers
#
# L - reset and begin of calling number
# D - continue of called number
#
ModemDialCmd:   ATDF%s  # user can override F by dial V
#ModemDialCmd:  ATDV%s  # user can override V by dial F
#ModemDialCmd:  ATD%sF  # user can't override F
#ModemDialCmd:  ATD%sV  # user can't override V
#ModemDialCmd:  ATD%sVL # user can't override V or calling 
number


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Re: [asterisk-users] pjsip user_eq_phone adds user=phone to anonymous user bug?

[Daniel Tryba]

On Fri, Jun 09, 2017 at 11:40:01AM -0300, Joshua Colp wrote:
> What seems to be happening is that the session is being set up and the
> user=phone parameter added. It's only after that the values are updated
> to be Anonymous and the user=phone parameter is left there. Please file
> an issue[1] with the description above.

Issue created:
https://issues.asterisk.org/jira/browse/ASTERISK-27047


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[asterisk-users] pjsip user_eq_phone adds user=phone to anonymous user bug?

[Daniel Tryba]

With pjsip (asterisk 13.14.1) I see the problem that an anonymous from
header gets user=phone appendend to the URI if user_eq_phone=yes is
specified:

On the incoming leg:
From: anonymous 
;tag=Q5zBj7BMnvI6Fe6O2866fox3ZHmn-smt
Get transformed to 
From: "Anonymous" 
;tag=fa3cb748-6af9-485f-8a70-a2b9ad40b13a
on the outgoing leg.

Setting user_eq_phone = no will result in user=phone not being added.
The upstream provide demands user=phone in URIs if the username
resembles a phonenumber, but declines the INVITE if user=phone is
present on an anonymous username.

Looking at the code,res/res_pjsip.c function ast_sip_add_usereqphone is
the only place I see that might add user=phone:

=
int i = 0;
//.
if (pj_strbuf(_uri->user)[0] == '+') {
i = 1;
}

/* Test URI user against allowed characters in AST_DIGIT_ANY */
for (; i < pj_strlen(_uri->user); i++) {
if (!strchr(AST_DIGIT_ANYNUM, pj_strbuf(_uri->user)[i])) {
break;
}
}

if (i < pj_strlen(_uri->user)) {
return;
}

  //add user=phone if we get to the code below
=

sip_uri->user should be "anonymous"
AST_DIGIT_ANY is: #define AST_DIGIT_ANYNUM "0123456789"

So in the for loop the first char of sip_uri->user should result in a
NULL from strchr. Leaving i at the value 0, which is smaller than the
length of sip_uri->user. And thus the function should return before
adding the user=phone. So why is user=phone being added?

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Re: [asterisk-users] asterisk server - no sound

[Daniel Tryba]

On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote:
> extensions.conf:
> [home]
> exten = 102,1,Answer()
> same =  n,Wait(1)

If this is copy and paste, then your dialplan is broken (= should be =>)

But to debug, enable logging (core set verbose 5), when needed debugging
(core set debug 5) and sip logging (sip set debug on / pjsip set logger
on).

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Re: [asterisk-users] OT: DMARC enabled domains on this list

[Daniel Tryba]

On Tue, Jun 06, 2017 at 08:23:33AM -0400, James B. Byrne wrote:
> > The reports are there to tell you something isn't right (like on this
> > mailing list). Disabling them is only hiding the problem, people might
> > be replying with the correct answer to a problem, but the OP might
> > never gets that message.
> >
> 
> What DMARC reports is that somebody other than yourself is sending
> email claiming to be you.  And there is absolutely nothing that you
> can do about it.  So the question arises: What is the value in these
> reports?

To tell those others (in the case of legitimate mail via mailinglists)
they are doing something wrong and mail redirected by said mailinglists
isn't getting delivered (or like with gmail "marked as phishing and put
into quarantine").

Also with increased use of DMARC (which I don't personally care for but
the BIG mail operators are kind of forcing it) there will be more
bounces from DMARCed senders to subscribed users which may result in the
mailinglist software to incorrectly unsubcribe those recipients.

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Re: [asterisk-users] Extensions of sip trunk

[Daniel Tryba]

On Tue, Jun 06, 2017 at 12:40:21AM +0200, Hans-Peter Jansen wrote:
> > Yes, something like if they can't fix the R-URI:
> > exten => X_.,n,Set(TO=${CUT(SIP_HEADER(To),@,1)})
> > exten => X_.,n,Set(TO=${CUT(TO,:,2)})
> > exten => X_.,n,Goto(somewhereelsetopreventloops${TO},1)
> 
> Sorry for the silly question, but how do I feed the TO variable back to the 
> usual pattern matching? Assign to $EXTEN?

The goto does that (with fixed typo):
exten => X_.,n,Goto(somewhereelsetopreventloops,${TO},1)
 
The reason to send to another context  where you handle specific DIDs is
to prevent loops. _X. is kind of a wildcard, if there is no DID for the
TO this will loop in _X.


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Re: [asterisk-users] Extensions of sip trunk

[Daniel Tryba]

On Mon, Jun 05, 2017 at 06:10:50PM +0200, Hans-Peter Jansen wrote:
> ; matches 12345678099, too
> exten => _1234567800,1,Dial(SIP/int)
> 
> Except from SIP invite with tcpdump:
> 
> INVITE sip:12345678@provider:5060 SIP/2.0
> From: ;tag=as6bc7cbbc
> To: 

12345678099 doesn't match _1234567800. The problem is the other side is
setting the R-URI to sip:12345678@provider for any number, so the
EXTEN matched in the dialplan is 12345678. Ask them to fix this
problem.

> I wonder, if I really need to grab the extension with 
> Set(DN=${SIP_HEADER(TO):5}) or something similar?

Yes, something like if they can't fix the R-URI:
exten => X_.,n,Set(TO=${CUT(SIP_HEADER(To),@,1)})
exten => X_.,n,Set(TO=${CUT(TO,:,2)})
exten => X_.,n,Goto(somewhereelsetopreventloops${TO},1)

> Another issue is, that I don't like asterisk to decline foreign INVITE 
> requests. Any best practices from within asterisk on how to ignore SIP 
> invitations, that don't match certain criteria (neither local nor from sip 
> provider)?

Don't enable guest access, or send any unknown/guest to a context that
will just hangup.

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Re: [asterisk-users] OT: DMARC enabled domains on this list

[Daniel Tryba]

On Mon, Jun 05, 2017 at 01:08:17PM -0400, James B. Byrne wrote:
> This is likely the issue surrounding mailing lists rewriting headers
> and/or modifying messages bodies or simply re-transmitting messages as
> the original sender from an unapproved domain. This was discussed at
> length on the ITEF mailing list.  Without seeing your headers and
> those of a recipient it is impossible to be sure but my spidy sense
> tells me this is so.

Subjects (atleast) are being rewritten, a recipient can't verify the
original (signed) hash to match the received message (replay
protection).  Only thing that is needed is a valid DKIM signature after
the subject (and maybe others) has "[asterisk-users]" prepended.

It appears exim 4.76 is being used, that version is recent enough to add
DKIM on sending via smtp.

begin transports

remote_smtp:
driver  = smtp
dkim_domain = lists.digium.com
dkim_selector   = auniqueid
dkim_private_key= /etc/exim4/dkim/list.digium.com-private.pem
dkim_canon  = relaxed

More info for example from:
https://debian-administration.org/article/718/DKIM-signing_outgoing_mail_with_exim4
The hints to do this for only 1 domain if the smtpd is used for others
are all there.
 
> You can manage this in your DNS forward zone by turning off the DMARC
> reporting request. No, I no longer recall the details.  Or you can
> simply direct the incoming reports to /dev/null.

The reports are there to tell you something isn't right (like on this
mailing list). Disabling them is only hiding the problem, people might
be replying with the correct answer to a problem, but the OP might never
gets that message.


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[asterisk-users] OT: DMARC enabled domains on this list

[Daniel Tryba]

Having enabled a strict DMARC setup I noticed everytime I send a message
here I get all these reports of messages which fail DMARC. Since I don't
want people to miss my wise thoughts maybe the maintainers of this list
could look into DKIM signing (or any of the other ways to work around
spf and dmarc breaking forwards)


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Re: [asterisk-users] Let's encrypt privkey : Specified certificate file could not be used

[Daniel Tryba]

On Fri, Jun 02, 2017 at 02:36:38PM +0200, Jonas Kellens wrote:
> [Jun  2 14:29:28]   == DTLS ECDH initialized (secp256r1), faster PFS enabled
> [Jun  2 14:29:28] ERROR[27360][C-0ae5]: res_rtp_asterisk.c:1441
> ast_rtp_dtls_set_configuration: Specified certificate file
> '/etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' for RTP instance
> '0x7f920c538a78' could not be used

What size is the privatekey? There is a script to create cert for
asterisk:
https://github.com/asterisk/asterisk/blob/master/contrib/scripts/ast_tls_cert
It create a 1024b keypair, maybe for a good reason. Certbot its size is
2048 by default. Try adding --rsa-key-size 1024 (our signing a
"handcrafted" key)


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Re: [asterisk-users] Forward error code beetwen legs

[Daniel Tryba]

On Thu, Jun 01, 2017 at 09:06:25PM +0200, Loic Chabert wrote:
> [gotoexternal]
> exten => _X.,1,Dial(SIP/${EXTEN}@provider)
> 
> When my SIP provider return to asterisk a 404 SIP error code, asterisk
> return to phone a 503 error code.
> 
> Why 503 error code has been returned, and not the original error = 404
> error ?
> 
> Asterisk's cli said: == Everyone is busy/congested at this time (1:0/0/1)
> 
> How can i return the 404 error ? Forward this 404 error to the original leg
> ?

To lazy to look it up exactly, but add something like:
exten => _X.,n,Hangup(${HANGUPCAUSE})


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

[Daniel Tryba]

On Wed, May 31, 2017 at 01:39:25PM -0700, Steve Edwards wrote:
> >What bugs you about the output format?
> 
> It's been a while, but as I recollect, it included the date/timestamp in the
> file name of the 'ring buffer' which meant that each time the host was
> rebooted, dumpcap didn't know the files from the previous run should be
> deleted when they 'aged out.'

Solvable by by writing a cleanup script that deletes files over a
specific age, just a basic find in the daily crontab:
find /path/to/captures -type f -name 'pattern*' -mtime +X -exec rm {} \;


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

[Daniel Tryba]

On Wed, May 31, 2017 at 12:36:47PM -0700, Steve Edwards wrote:
> I want to capture all SIP messages.
> 
> I have about 30 hosts in about 6 colos.
> 
> My first thought was dumpcap, but the output file name format bugs me.
> 
> What do you use for long term SIP capture?

What bugs you about the output format? There are multiple ways to
display stored information, wireshark can be extremely usefull (and
unstable) or just dump plain text by replaying the pcap with ngrep.

Ways I used so far:

-tshark to produce pcap file (-b duration:x to split up files into time
 intervals
-"sip set log on" to store it plain text in asterisk log files (or pjsip
 set logger on)
-ngrep -W byline to store it in 

Will look into in the near future:
-Homer via res_hep_pjsip
-voipmonitor (didn't know about till just now thanks to Marks reply)

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Re: [asterisk-users] pjsip direct_media=yes and "unknown" endpoints

[Daniel Tryba]

On Wed, Apr 26, 2017 at 06:25:43PM +0200, Daniel Tryba wrote:
> Whoever when a terminating call comes in from the uplink provider, a
> sip request is send to a redirector. The redirector has
> redirect_method=uri_core configured (the only method that works for
> me).

[...]

> The request now gets routed based on a fully qualified domainname
> (with NAPTR/SRV records), which ultimately resolves to an ip that is
> matched in the endpoint SBC used above to originate a call.  But now
> the asterisk stays in the loop regarding RTP, a simple bridge is
> created but never switches to direct media.

This is not an asterisk problem, after fiddling with the config and
using templates to make sure the config for all (configured) endpoints
was the same, a reINVITE renegotiated RTP between the endpoints. 

However what happens is that after the renegotiation the DTMF
payloadtype (rfc4733) changed from 101 in the initial setup to 96. The
uplink provider doesn't support this thus DTMF breaks making
direct_media not an option right now. Something I have to figure out
later.

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Re: [asterisk-users] Dial an extension to modify dialplan

[Daniel Journo]

> Hello

> I have the following scenario:

> [mynicecontext]
> exten => 2000,1,Dial(SIP/deviceA/deviceB/deviceC)

> As expected, by dialing 2000, all three devices will ring. And that's fine.
> However, there are situations where I only want "deviceA" and "deviceB"
to ring. I would like to have an extension to dial in order to modify the 
dialplan.

> Is there a better solution?

Take a look at https://wiki.asterisk.org/wiki/display/AST/Device+State
Specifically, Custom Device states.

You write both versions of the dialplan, and use an IF on the custom device 
state to determine which one runs.
You can then dial 4000 to turn the Custom Device from Busy to Available to set 
which section of the dialplan to run.
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[asterisk-users] Feature Code to Meeting Room

[Daniel Journo]

Hi,

Is it possible to set up a feature code to move both a caller and callee to a 
meeting room?
If yes, what should I be looking at?

Bonus question, is it possible to then automatically dial a 3rd person and 
invite them to the meeting room?

The client wants to do this with the push of a couple of buttons only.

Many thanks
Dan

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Re: [asterisk-users] pjsip direct_media=yes and "unknown" endpoints

[Daniel Tryba]

> > Anybody got an idea why the last scenario fails to work?
> 
> If you turn up core debug (core set debug 2) and ensure it is going to
> the CLI then the bridge_native_rtp module will tell you why exactly it
> can't native bridge. You might also want to do a core show channel on
> both channels to see what the codecs are.

Thanks for the hint, I wasn't seeing any debug since it wasn't getting
send to console. I'll take a better look and report back.


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[asterisk-users] pjsip direct_media=yes and "unknown" endpoints

[Daniel Tryba]

I'm trying to implement direct_media between multiple peers and an
uplink provider, all of whom have direct_media=yes configures.

For originating calls to the uplink provider direct_media=yes works like
expected. SIP flows through asterisk, rtp doesn't

SIP: enduser <-> SBC <-> asterisk 13 <-> uplink
RTP: enduser <-> SBC <-> uplink

SBC matches an endpoint based on ip and dials the uplink:

-- Executing [+31x@outgoingrr:9] Dial("PJSIP/sbcs-0092", 
"PJSIP/+31x@uplink") in new stack
-- Called PJSIP/+31x@uplink
-- PJSIP/uplink-0093 is making progress passing it to PJSIP/sbcs-0092

-- PJSIP/uplink-0093 answered PJSIP/sbcs-0092
-- Channel PJSIP/uplink-0093 joined 'simple_bridge' basic-bridge 
<3b25c543-13a3-4d74-b2fe-7122a1cfe4a4>
-- Channel PJSIP/sbcs-0092 joined 'simple_bridge' basic-bridge 
<3b25c543-13a3-4d74-b2fe-7122a1cfe4a4>
   > Bridge 3b25c543-13a3-4d74-b2fe-7122a1cfe4a4: switching from simple_bridge 
technology to native_rtp
   > Remotely bridged 'PJSIP/sbcs-0092' and 'PJSIP/uplink-0093' - media 
will flow directly between them
   > Remotely bridged 'PJSIP/sbcs-0092' and 'PJSIP/uplink-0093' - media 
will flow directly between them

Whoever when a terminating call comes in from the uplink provider, a sip
request is send to a redirector. The redirector has redirect_method=uri_core
configured (the only method that works for me).

-- Executing [+31x@incoming:11] Dial("PJSIP/uplink-0094", 
"PJSIP/+31x@pathfinder") in new stack
-- Called PJSIP/+31x@pathfinder
-- Now forwarding PJSIP/uplink-0094 to 
'PJSIP/pathfinder/sip:+31xx...@sip.xx.nl' (thanks to 
PJSIP/pathfinder-0095)
...
-- PJSIP/pathfinder-0096 answered PJSIP/uplink-0094
-- Channel PJSIP/pathfinder-0096 joined 'simple_bridge' basic-bridge 
<1bf02059-ea8f-4f9c-bc33-8ae99ba45c9a>
-- Channel PJSIP/uplink-0094 joined 'simple_bridge' basic-bridge 
<1bf02059-ea8f-4f9c-bc33-8ae99ba45c9a>
...
-- Channel PJSIP/pathfinder-0096 left 'simple_bridge' basic-bridge 
<1bf02059-ea8f-4f9c-bc33-8ae99ba45c9a>
-- Channel PJSIP/uplink-0094 left 'simple_bridge' basic-bridge 
<1bf02059-ea8f-4f9c-bc33-8ae99ba45c9a>

The request now gets routed based on a fully qualified domainname (with
NAPTR/SRV records), which ultimately resolves to an ip that is matched in the
endpoint SBC used above to originate a call.  But now the asterisk stays in the
loop regarding RTP, a simple bridge is created but never switches to direct
media.

SIP: enduser <-> uplink <-> asterisk 13 <-> pathfinder (302 redirect)

SIP: enduser <-> uplink <-> asterisk 13 <-> sip.xx.nl
RTP: enduser <-> uplink <-> asterisk 13 <-> sip.xx.nl

Anybody got an idea why the last scenario fails to work?


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Re: [asterisk-users] best kernel for Asterisk

[Daniel-Constantin Mierla]

Hello,


On 19.04.17 09:57, marek cervenka wrote:
> hi,
>
> what kernel version are you using for asterisk?
>
> are you satisfied with distro kernel (centos 6 2.6.32, centos 7 3.10,
> ...) ?
>
> are you using newer kernels from elrepo.org?
>
> which kernel features are most critical for Asterisk performance pattern?
>
I prefer to work with the standard kernel from the distro (using debian)
for security reasons, unless there is a team in the company with very
good kernel tuning knowledge. Probably one can squeeze some more
performances with a custom kernel build, but in long time that typically
becomes a maintenance nightmare.

If needed, you can instead aim for horizontal scaling by deploying a
farm of Asterisk systems with a sip proxy load balancer in front of it
(well, I could be a bit biased, because I do work mostly with the
kamailio sip proxy). Anyhow, to cut it straight, standard distro kernel
worked fine for the deployments I was involved in.

Cheers,
Daniel

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

[Daniel Journo]

> Hello
> as you can read in my original post "moh reload" and "module reload 
> res_musiconhold.so" does nothing.
> Only at restart the new files are available.
> Is this a bug ?? How can I get more debugging for this problem ??

Just spotted that you are using Asterisk 1.8.32.3.
The bug I'm thinking of is in the latest version. I don't know if your version 
is affected.

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

[Daniel Journo]

> Hello
> as you can read in my original post "moh reload" and "module reload 
> res_musiconhold.so" does nothing.
> Only at restart the new files are available.
> Is this a bug ?? How can I get more debugging for this problem ??

I think there is currently a bug with MOH. For now, if you add a file to a moh 
folder, 'touch musiconhold.conf' and then reload moh.
Please let me know how it goes.

Kind regards

Dan Journo

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Re: [asterisk-users] Beep on Attended Transfer

[Daniel Journo]

Ø  Set the ATTENDED_TRANSFER_COMPLETE_SOUND channel variable to the sound
file to play on a transfer.
>Does that have to be set in the Dial handler in order to get set on the 
>dialled channel or can it be inheritted?

Never mind. Tested it. Working great! Thanks.
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Re: [asterisk-users] Beep on Attended Transfer

[Daniel Journo]

Ø  Set the ATTENDED_TRANSFER_COMPLETE_SOUND channel variable to the sound
file to play on a transfer.
Does that have to be set in the Dial handler in order to get set on the dialled 
channel or can it be inheritted?

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[asterisk-users] Beep on Attended Transfer

[Daniel Journo]

Hi,

During an attended transfer using the SIP phone feature buttons, I'm getting a 
few complaints from recipients that they can't tell when the call they are 
receiving has been transferred.
Is there any way (even if it's complicated) to generate a beep tone to the 
recipient of the transferred call when the transfer is completed?

I know you can do this with DTMF codes but they want to use the phone's 
transfer features.

Many thanks
Dan

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[asterisk-users] Asterisk versions supporting Path header?

[Daniel Pocock]

The bug tracker includes several issues relating to Path (RFC 3327)
support.  It is not clear which version actually included the patch and
which versions are working.

Could anybody update these issues in Jira with a brief comment about the
supported versions?


https://issues.asterisk.org/jira/browse/ASTERISK-16884
   original patch against chan_sip / Asterisk 1.8
   Status is "Fixed", but not version is recorded,
   which version was this merged in?

https://issues.asterisk.org/jira/browse/ASTERISK-21084
   chan_pjsip Path support
   Satus is Fixed for v12.1.0
   - is that only for chan_pjsip, or is Path also
 supported in chan_sip in any versions up to 12.1.0?

https://issues.asterisk.org/jira/browse/ASTERISK-25666
   Path header ignored (looks like a regression?)
   reported for 13.6.0 - which is the last version where it did work?


https://jira.digium.com/browse/SWP-2484
   "add Path header support to chan_sip"
   Internal Jira link - does this issue contain any further
   details about the versions supported?

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[asterisk-users] Trouble getting Asterisk Running with FreePBX 11

[Daniel Chavez]

Hi,
I recently had to reinstall Asterisk and FreePBX. asteirsk 11.20 and FreePBX 12.
This is running on Centos 6.7 32 bit.
When I use amportal start
It comes up with the errors below
Error in argument 1, char 2: option not found r
/usr/local/sbin/amportal: line 49: Usage:: command not found
 WARNING: ERROR IN CONFIGURATION 
astrundir in '/etc/asterisk' is set to  but the directory
does not exists. Attempting to create it with: 'mkdir -p '
mkdir: missing operand
Try `mkdir --help' for more information.
 ERROR: COULD NOT CREATE  
Attempt to execute 'mkdir -p ' failed with an exit code of 1
You must create this directory and the try again.
I'm not sure how to solve this? Asterisk already has the run directory in 
/var/run/asterisk and the config file notes this.
Perhaps it's PHP? Before I ran PHP 5.4 and now am running PHP 5.5.
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Re: [asterisk-users] Queues - periodic announce while ringing members

[Daniel Chavez]

Ish,
I use the same version of Asterisk on CentOS 6.7. I wonder the same thing. 
Hopefully we will find this out.


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Re: [asterisk-users] Voice recognition IVR Is it possible?

[Daniel Heckl]

Read README, check the requirements and get the google speech api key.

Then add a custom destination in FreePBX and edit your extensions_custom.conf.

> Am 22.02.2016 um 21:03 schrieb Daniel Chavez <topdog2...@gmail.com>:
> 
> Ok. Where I am now is, I copied the speech-recog.agi to the agi-bin directory.
> Where do I go from here?
> 
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Re: [asterisk-users] Voice recognition IVR Is it possible?

[Daniel Chavez]

Ok. Where I am now is, I copied the speech-recog.agi to the agi-bin directory.
Where do I go from here?


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Re: [asterisk-users] Voice recognition IVR Is it possible?

[Daniel Heckl]

I use FreePBX as well. There is no module for speech recognition. You have too 
create a custom destination.

> Am 22.02.2016 um 20:53 schrieb Daniel Chavez <topdog2...@gmail.com>:
> 
> Thanks, this looks promising. I was wondering if there's an easier way to get 
> this to work inside FreePBX?
> I have all of the dependencies installed for it, but now I want to know if 
> there's a mod I can use in FreePBX to get it setup?
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Re: [asterisk-users] Voice recognition IVR Is it possible?

[Daniel Chavez]

Thanks, this looks promising. I was wondering if there's an easier way to get 
this to work inside FreePBX?
I have all of the dependencies installed for it, but now I want to know if 
there's a mod I can use in FreePBX to get it setup?
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Re: [asterisk-users] Voice recognition IVR Is it possible?

[Daniel Heckl]

Daniel,

try this http://zaf.github.io/asterisk-speech-recog/.

I have tested it myself, it works very well.

Daniel

> Am 22.02.2016 um 19:34 schrieb Daniel Chavez <topdog2...@gmail.com>:
> 
> Thanks for the link.
> Are there no free alternatives for speech recognition?
> 
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Re: [asterisk-users] Voice recognition IVR Is it possible?

[Daniel Chavez]

Thanks for the link.
Are there no free alternatives for speech recognition?

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[asterisk-users] Voice recognition IVR Is it possible?

[Daniel Chavez]

Hello list,
I was wondering if it were possible for asterisk to do a voice recognition type 
IVR?
Like you know how most banks and stuff do, where they ask you to say your 
selection or key it in?
If this is possible, how can I set this up? I'm using FreePBX 2.11 on Linux, 
CentOS 6.7 32-bit, asterisk version 1.8.32.3.
Thanks.
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Re: [asterisk-users] 1000 analogue lines with asterisk

[Daniel Harper]

What about leaving the old PBX in place and trunking it via ISDN to the
asterisk server.

We use rhino 24 channel bank but are 2U for rhino + 1U for patch panel.
(RJ21 cable so might be able to use existing ones if they are RJ21)

Used USB xorcoms a while back, things may of changed but if one is down and
reboot server then asterisk doesn't come up.

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Re: [asterisk-users] Nube question: where is chan_sip.so?

[Daniel Chavez]

Hello,
My name is Daniel.
Very strange. Chan_Sip.so should be there. Maybe the cat didn't have enough cat 
food? *smile*
I am a System Administrator, and have done plenty of Asterisk install's, so am 
use to errors and such.
If you'd like, I can remote SSH in to troubleshoot your install. I charge 
$25USD for this, but will wave said fea since you're new to Asterisk.
If you'd like help, please email off-list to
webmas...@firestar-hosting.com <mailto:webmas...@firestar-hosting.com>
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Re: [asterisk-users] PJSIP Stun/ICE

[Daniel Heckl]

Bryant,

I have the same problem with dynamic public IPs and PJSIP. What is your idea to 
solve the problem?

My suggestion would be to write a script that monitors the change, 
pjsip.transports.conf updated and Asterisk restarts?

Daniel

> Am 26.01.2016 um 14:21 schrieb Joshua Colp <jc...@digium.com>:
> 
> Bryant Zimmerman wrote:
>> Joshua
>> So once a transport is pulled from the transports table in realtime
>> during asterisk startup it can't get any updates?
>> Can a new transport be added to the table and the associated endpoints
>> be updated to use the new transport, or are transport types only read at
>> startup across the board?
> 
> Transports can only be loaded at startup. This stems from PJSIP not being 
> dynamic with transports (it doesn't like its environment changed to that 
> degree while in use). I'm afraid if your IP changes you'd have to restart 
> Asterisk when you are using PJSIP.
> 
> -- 
> 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] PJSIP Stun/ICE

[Daniel Heckl]

Bryant,

that sounds interesting. I am searching for a script which monitors and updates 
the ip address. Does this your script? Can you share your script with us?

Thanks
Daniel

> Am 26.01.2016 um 16:39 schrieb Bryant Zimmerman <brya...@zktech.com>:
> 
> Daniel
>  
> Thank you for your response. I was considering this as well. I have a script 
> that monitors the IP Address now. I was hoping to use the real-time 
> transports table now that alembic creates. I am trying to figure out which 
> pjsip module is responsible for the transports contexts as I need to now 
> configure it in the sorcery.conf file. I thought it would be under the 
> [res_pjsip] context, but it is not even trying to pull from my transports 
> table when it is there.  I am hoping someone will know what module it is in 
> so I can move my configuration under the correct context.
>  
> Thanks
> 
> Bryant
>  
> From: "Daniel Heckl" <daniel.he...@gmail.com>
> Sent: Tuesday, January 26, 2016 10:15 AM
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> <asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] PJSIP Stun/ICE
>  
> Bryant,
> 
> I have the same problem with dynamic public IPs and PJSIP. What is your idea 
> to solve the problem?
> 
> My suggestion would be to write a script that monitors the change, 
> pjsip.transports.conf updated and Asterisk restarts?
> 
> Daniel
> 
> > Am 26.01.2016 um 14:21 schrieb Joshua Colp <jc...@digium.com>:
> >
> > Bryant Zimmerman wrote:
> >> Joshua
> >> So once a transport is pulled from the transports table in realtime
> >> during asterisk startup it can't get any updates?
> >> Can a new transport be added to the table and the associated endpoints
> >> be updated to use the new transport, or are transport types only read at
> >> startup across the board?
> >
> > Transports can only be loaded at startup. This stems from PJSIP not being 
> > dynamic with transports (it doesn't like its environment changed to that 
> > degree while in use). I'm afraid if your IP changes you'd have to restart 
> > Asterisk when you are using PJSIP.
> >
> > --
> > 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] Transfer calls "on demand"

[Daniel Heckl]

You are searching for „Call Pickup“. It is implemented in Asterisk by default.

https://wiki.asterisk.org/wiki/display/AST/Call+Pickup 
<https://wiki.asterisk.org/wiki/display/AST/Call+Pickup>
Take a look under section „Configuration Options“.

Daniel

> Am 29.12.2015 um 07:53 schrieb Luca Bertoncello <lucab...@lucabert.de>:
> 
> Hi list!
> 
> Right now I configured my Asterisk to forward the calls for the number X to
> both phones (mine and the phone of my wife).
> It works, of course, but I'm not enthusiast...
> 
> I see what we have at office: if one phone rings, other phones in the same
> group can "catch the call", so that if a colleague is not present, another
> colleague can catch the call.
> 
> I'd like to have the same procedure at home. I think, Asterisk can do that,
> but I have no idea how to implement this.
> 
> Shortly: what I want is that every phone rings only on calls for the own
> number, and I can catch the call from the other phone, if for example my wife
> is not at home, for example pressing "*5#" or other key combination.
> 
> Thanks a lot for your suggestion!
> 
> Luca Bertoncello
> (lucab...@lucabert.de)
> 
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Re: [asterisk-users] Transfer calls "on demand"

[Daniel Heckl]

On top of the page: "Call pickup support added in Asterisk 11“

I think that is the problem. I do not know a solution for 1.8, but maybe 
someone other.

> Am 29.12.2015 um 10:20 schrieb Luca Bertoncello <lucab...@lucabert.de>:
> 
> Daniel Heckl <daniel.he...@gmail.com> schrieb:
> 
>> You are searching for „Call Pickup“. It is implemented in Asterisk by
>> default.
>> 
>> https://wiki.asterisk.org/wiki/display/AST/Call+Pickup
>> <https://wiki.asterisk.org/wiki/display/AST/Call+Pickup> Take a look under
>> section „Configuration Options“.
> 
> Hi, Daniel!
> 
> Thanks for your answer...
> I'm using Asterisk 1.8.30.0 on an OpenWRT-Router.
> I found the configuration for call pickup in the sip.conf and features.conf,
> so I tried to activate it...
> Unfortunately, unsuccessfully...
> 
> So, my sip.conf:
> 
> callgroup=1,3-4 ; We are in caller groups 1,3,4
> pickupgroup=1,3-5   ; We can do call pick-p for call group 1,3,4,5
> 
> my features.conf:
> 
> ; Pickup Options
> ;
> pickupexten = *8   ; Configure the pickup extension. (default is 
> *8)
> ;pickupsound = beep ; to indicate a successful pickup (default: 
> no sound)
> ;pickupfailsound = beeperr  ; to indicate that the pickup failed 
> (default: no sound)
> 
> my users.conf:
> 
> [general]
> callgroup = 1
> pickupgroup = 1
> 
> my extensions.conf:
> 
> [anika_incoming]
> exten => _0049351222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
> exten => _0049351222,Set(CHANNEL(pickupgroup)=1)
> exten => _0049351222,n,Dial(local/222@anika_incoming)
> exten => _0351222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
> exten => _0351222,n,Dial(local/222@anika_incoming)
> exten => _222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
> exten => _222,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" = "+49" 
> ]?0${CALLERID(num):3}:${CALLERID(num)})})  ; Damit das "+49" mit "0" ersetzt 
> wird
> exten => _222,n,Set(CHANNEL(musicclass)=default)
> ;;;exten => 
> _222,n,Dial(SIP/0049351222/1@luca_for_anika_voip_mobile,19,RcxX)
> exten => _222,n,Dial(SIP/0049351222,19,RcxX)
> exten => _222,n,Verbose(2,Voicemail for Anika)
> exten => _222,n,Set(CALLERID(name)=)  
>  ; Damit in der E-Mail der AB nicht den Namen steht
> exten => _222,n,VoiceMail(0049351222,us)
> exten => _222,n,Hangup
> 
> Then I called the 222 with my mobile phone and I tried to get the call
> from the other phone, calling the *8.
> Unfortunately I get an error (invalid number) on the display of the phone,
> and the phone 222 continue to ring.
> No error on the log of Asterisk...
> 
> Any suggestion?
> 
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
> 
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