[asterisk-users] Setting PJSIP header from AMI

[Alex Zarubin]

Hello,

We are moving from an older asterisk/SIP to a newer (18+) asterisk/PJSIP and 
trying to figure out how to add [identity] header when originating a call from 
AMI/PAMI.
In the older version we would just set a variable like this:

$action = new OriginateAction("SIP/");
$action->setVariable('__SIPADDHEADER51',"Identity: $identity");  // 
$identity contains generated by 3rd party header

Is there anything similar for

$action = new OriginateAction("PJSIP/");
???

that would work for PJSIP?

Any suggestions are appreciated.

Alex Zarubin/TTH




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Re: [asterisk-users] [External] 180 Ringing missing

[Alex Hermann]

On dinsdag 1 december 2020 13:22:17 CET Floimair Florian wrote:
> If you have 183 Session progress, there is no need to send 180 Ringing
> (especially not AFTER 183 Session progress), as you already have
> early media instead. Having both is actually a bit misleading IMHO.

I disagree (and I think rfc3261 agrees).  How would the caller know if 
the callee has been alerted if he doesn't receive the 180?

183 and 180 have different meaning. 180 indicates the callee is being 
alerted. An 183 has no such meaning but is often used to setup early 
media (although any 1xx can do that). 180 and 183 are not mutually 
exclusive and in fact form a full matrix of possible and useful states. 
There can be early media without any phone ringing (announcement), 
ringing without early media (caller is expected to generate a ringback 
tone itself), no early media nor ringing and finally early media and 
ringing simultaneous.

Especially automata calling will want to know the difference.

Of course, a generated ringback tone by the caller should be stopped 
when media is received.

Asterisk should indicate to the caller the same state it received from 
the callee.
-- 
Alex Hermann



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Re: [asterisk-users] Redis in place of astdb

[alex epshteyn]

I’ll second that - for CDR you want the fastest sequential writing with 
possible batching of CDR records

Best regards,
Alex

www.thirdlane.com



> On Jul 9, 2020, at 1:37 AM, Antony Stone 
>  wrote:
> 
> On Thursday 09 July 2020 at 00:50:28, Jon Bonilla (Manwe) wrote:
> 
>> DO you know odbc redis drivers? It would be nice to store cdrs ans other
>> stuff in redis without patching asterisk
> 
> A quick Google search turns up 
> https://www.cdata.com/kb/tech/redis-odbc-python-linux.rst
> which I have no experience of and cannot comment on, but looks like what you 
> need.
> 
> PS: I question the wisdom of storing CDRs in Redis - I think an RDBMS is the 
> correct tool for *that* job.  I agree that Redis may be useful in other 
> areas, 
> though.
> 
> 
> Antony.
> 
> -- 
> All matter in the Universe can be placed into one of two categories:
> 
> 1. Things which need to be fixed.
> 2. Things which need to be fixed once you've had a few minutes to play with 
> them.
> 
>   Please reply to the list;
> please *don't* CC me.
> 
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> 
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Re: [asterisk-users] WebRTC as Softphone substitute ?

[alex epshteyn]

No, this will not be a problem. TURN relays media based on RTP headers which 
are unencrypted.  SRTP encrypts only the payload of RTP packets. DTLS-SRTP is 
one of the things mandatory to implement and support things in WebRTC, and it 
will not speak RTP/AVP. Asterisk may have to unencrypt payload for media paths 
that require that, which TURN server is not able to do. 

Best regards,
Alex

Alex Epshteyn
a...@thirdlane.com
+1 (415) 261 6601
www.thirdlane.com



> On Oct 2, 2018, at 7:39 PM, David P  wrote:
> 
> Thanks for sharing this, Alex. It sounds like TURN, as a media repeater, 
> wouldn't work if the media must be secured (via SRTP). Is that right?
> 
> On Wed, 3 Oct 2018, 3:17 pm alex epshteyn,  <mailto:a...@thirdlane.com>> wrote:
> WebRTC requires a few specific things to be in place. We have blog posts that 
> talk about WebRTC based Thirdlane Connect, but most of the information 
> applies to WebRTC applications in general.
> 
> https://www.thirdlane.com/blog/get-up-and-running-with-thirdlane-connect 
> <https://www.thirdlane.com/blog/get-up-and-running-with-thirdlane-connect>
> 
> https://www.thirdlane.com/blog/nat-stun-turn-and-ice 
> <https://www.thirdlane.com/blog/nat-stun-turn-and-ice>
> 
> Best regards,
> Alex
> 
> 
> Alex Epshteyn
> a...@thirdlane.com <mailto:a...@thirdlane.com>
> +1 (415) 261 6601
> www.thirdlane.com <http://www.thirdlane.com/>
> 
> 
> 
>> On Oct 2, 2018, at 6:08 PM, Nasir Iqbal > <mailto:na...@ictinnovations.com>> wrote:
>> 
>> @Olivior
>> I agree that seting up WebRTC is hard, however when done it is smooth to 
>> use. For replication you can build RPMs with working configurations.
>> 
>> Regarding stability, it is not being used widly, so can't say it is mature. 
>> However we have no complain so far regarding audio or connectivity. sometime 
>> we provide support for "allow media / mic" type issues, but you know it is 
>> security feature and not a bug.
>> 
>> Regards
>> 
>> On Tue, Oct 2, 2018, 13:03 Olivier > <mailto:oza.4...@gmail.com>> wrote:
>> @Nasir:
>> Thanks for replying here.
>> 
>> Did you met in your deployments, the kind of stability issues Carlos 
>> reported earlier ?
>> 
>> Le sam. 29 sept. 2018 à 13:32, Nasir Iqbal > <mailto:na...@ictinnovations.com>> a écrit :
>> Hi Olivior,
>> 
>> We have recently worked on a WebRTC based agent panel. As based on my 
>> experience I think that WebRTC based phones are far better and cheaper then 
>> those soft / sip phone. the big plus is that they are easy to customize and 
>> developer can use the power of browser and web to build / offer features 
>> which are not possible with regular phones. 
>> 
>> Regarding your concern about BLF or call history, for me as a being 
>> developer it is just a matter of customization.
>> 
>> Regards
>> 
>> Nasir Iqbal
>> 
>> ICTBroadcast - an Auto Dialer software for ITSP 
>> <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition>
>> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns
>> http://www.ictbroadcast.com/ <http://www.ictbroadcast.com/>
>> 
>> 
>> On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez > <mailto:cur...@telecomab.mx>> wrote:
>> On 9/26/18 10:20 AM, Matthew Fredrickson wrote:
>> 
>> > On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez > > <mailto:cur...@telecomab.mx>> wrote:
>> >> On 9/26/2018 4:46 AM, Olivier wrote:
>> >>
>> >>> Hello,
>> >>>
>> >>> This morning, I asked myself if WebRTC could be a viable alternative
>> >>> to softphone deployment.
>> >>>
>> >>> For me, main issue with Softphones is the amount of work needed for
>> >>> installation and configuration.
>> >>> Also, Softphones must be carefully choosen if Deskphone-like quality
>> >>> is expected.
>> >>>
>> >>> Now that WebRTC becomes ubiquitous, it might make sense to trade
>> >>> Softphone features (call history, BLF, ...) for WebRTC deployment
>> >>> simplicity.
>> >>>
>> >>> What do you think of this ?
>> >>> What kind of experience did you met with such WebRTC deployments ?
>> >>> What about classic telephony features (CallTransfer) ?
>> >>> Have you tried Cyber Maga Phone 2K ?
>> >>>
>> >>   If you can get it to work WebRTC is a goo

Re: [asterisk-users] WebRTC as Softphone substitute ?

[alex epshteyn]

WebRTC requires a few specific things to be in place. We have blog posts that 
talk about WebRTC based Thirdlane Connect, but most of the information applies 
to WebRTC applications in general.

https://www.thirdlane.com/blog/get-up-and-running-with-thirdlane-connect

https://www.thirdlane.com/blog/nat-stun-turn-and-ice

Best regards,
Alex


Alex Epshteyn
a...@thirdlane.com
+1 (415) 261 6601
www.thirdlane.com



> On Oct 2, 2018, at 6:08 PM, Nasir Iqbal  wrote:
> 
> @Olivior
> I agree that seting up WebRTC is hard, however when done it is smooth to use. 
> For replication you can build RPMs with working configurations.
> 
> Regarding stability, it is not being used widly, so can't say it is mature. 
> However we have no complain so far regarding audio or connectivity. sometime 
> we provide support for "allow media / mic" type issues, but you know it is 
> security feature and not a bug.
> 
> Regards
> 
> On Tue, Oct 2, 2018, 13:03 Olivier  <mailto:oza.4...@gmail.com>> wrote:
> @Nasir:
> Thanks for replying here.
> 
> Did you met in your deployments, the kind of stability issues Carlos reported 
> earlier ?
> 
> Le sam. 29 sept. 2018 à 13:32, Nasir Iqbal  <mailto:na...@ictinnovations.com>> a écrit :
> Hi Olivior,
> 
> We have recently worked on a WebRTC based agent panel. As based on my 
> experience I think that WebRTC based phones are far better and cheaper then 
> those soft / sip phone. the big plus is that they are easy to customize and 
> developer can use the power of browser and web to build / offer features 
> which are not possible with regular phones. 
> 
> Regarding your concern about BLF or call history, for me as a being developer 
> it is just a matter of customization.
> 
> Regards
> 
> Nasir Iqbal
> 
> ICTBroadcast - an Auto Dialer software for ITSP 
> <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition>
> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns
> http://www.ictbroadcast.com/ <http://www.ictbroadcast.com/>
> 
> 
> On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez  <mailto:cur...@telecomab.mx>> wrote:
> On 9/26/18 10:20 AM, Matthew Fredrickson wrote:
> 
> > On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez  > <mailto:cur...@telecomab.mx>> wrote:
> >> On 9/26/2018 4:46 AM, Olivier wrote:
> >>
> >>> Hello,
> >>>
> >>> This morning, I asked myself if WebRTC could be a viable alternative
> >>> to softphone deployment.
> >>>
> >>> For me, main issue with Softphones is the amount of work needed for
> >>> installation and configuration.
> >>> Also, Softphones must be carefully choosen if Deskphone-like quality
> >>> is expected.
> >>>
> >>> Now that WebRTC becomes ubiquitous, it might make sense to trade
> >>> Softphone features (call history, BLF, ...) for WebRTC deployment
> >>> simplicity.
> >>>
> >>> What do you think of this ?
> >>> What kind of experience did you met with such WebRTC deployments ?
> >>> What about classic telephony features (CallTransfer) ?
> >>> Have you tried Cyber Maga Phone 2K ?
> >>>
> >>   If you can get it to work WebRTC is a good option.  The problem is
> >> that any changes in your network may disrupt it and even trying to
> >> replicate your installation is difficult.  I have it working fine on my
> >> website so customers can call us directly from our web page but I never
> >> could get Cyber Mega Phone 2K to work on the same server.  We used JSSIP
> >> to create the webrtc phone on our website.
> > We just updated the documentation for how to get CMP2K working on the
> > wiki [1].  We'd love some feedback if you still have issues getting it
> > setup so that we can improve the docs.
> >
> > [1] 
> > https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone
> >  
> > <https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone>
> >
> > Best wishes,
> > Matthew Fredrickson
> >
>  I followed the procedure indicated in the link but I cannot get 
> remote video.  I can only see my own feed.  We do have audio for a 
> little while.  For some reason the users get disconnected after a few 
> minutes even though you can still see your video feed on screen.  This 
> was done with Asterisk 15.6.0
> 
> -- 
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> +52 (55)8116-9161
> 
> 
> -- 
> ___

[asterisk-users] Asterisk kafka connector feedback

[Alex Pappas]

Dear all,

I recently developed a connector
<https://github.com/gnostix/kafka-connect-asterisk> for Kafka and Asterisk
AMI.

The purpose is to send Asterisk AMI events, e.g. CDRs to a Kafka topic.
This could have many uses, like running billing or analytics through Kafka
streams in real time.

Please have a look and let me know if you have any feedback or use cases.

Cheers,
Alex
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Re: [asterisk-users] Dial to FastAGI application appears as 1-second CDR - how do I fix?

[Alex Villací­s Lasso]

El 29/05/18 a las 05:24, Tony Mountifield escribió:

In article <3a005ff6-19a4-215b-4751-bee616ec7...@palosanto.com>,
Alex Villací­s Lasso  wrote:

In my application, I am using AMI to run an Originate command between a channel 
and a dialplan application (NOT a
context). In my case, the application I want to invoke is FastAGI. The 
Originate AMI command works correctly, but
Asterisk generates a very
short (0-1s) duration for the CDR that results from this call, regardless of 
the time spent running the FastAGI
application. I want the CDR duration to reflect the time spent in the entirety 
of the call, as is normal for ordinary
calls. What should I do? I
am running Asterisk 11.25.3 .

Are you using Local channels as the target channel?

Yes, this is the standard way my application works.


I found what I believe is an issue with CDR handling when masquerading Local 
channels, when I ported
one of my old applications from Asterisk 1.2.32 to 11.25.3. It was recording an 
incorrect CDR
for the setup part of the call, and none for the answered part. In 1.2.32 I was 
getting two CDRs,
one for the setup part and another for the answered part.

I eventually tracked it down to ast_do_masquerade() in channel.c, and made the 
following change,
which corrected the CDR behaviour back to the same as what Asterisk 1.2 had 
done:

$ diff -u --show-c-function channel.c.orig channel.c
--- channel.c.orig  2017-09-19 17:03:38.0 +0100
+++ channel.c   2018-05-29 11:13:41.0 +0100
@@ -6864,7 +6864,9 @@ int ast_do_masquerade(struct ast_channel
 } exchange;
 struct ast_channel *clonechan, *chans[2];
 struct ast_channel *bridged;
+#ifdef I_THINK_THIS_IS_WRONG /* Tony Mountifield, 2018-03-29. Removing this 
code fixes lost CDRs with masquerade */
 struct ast_cdr *cdr;
+#endif
 struct ast_datastore *xfer_ds;
 struct xfer_masquerade_ds *xfer_colp;
 struct ast_format rformat;
@@ -7035,10 +7037,12 @@ int ast_do_masquerade(struct ast_channel
 ast_channel_tech_pvt_set(original, ast_channel_tech_pvt(clonechan));
 ast_channel_tech_pvt_set(clonechan, t_pvt);

+#ifdef I_THINK_THIS_IS_WRONG /* Tony Mountifield, 2018-03-29. Removing this 
code fixes lost CDRs with masquerade */
 /* Swap the cdrs */
 cdr = ast_channel_cdr(original);
 ast_channel_cdr_set(original, ast_channel_cdr(clonechan));
 ast_channel_cdr_set(clonechan, cdr);
+#endif

 /* Swap the alertpipes */
 ast_channel_internal_alertpipe_swap(original, clonechan);
$

I didn't research what version this change appeared in, nor whether it has been 
subsequently fixed in
later versions of Asterisk. There was no point reporting it in Asterisk 11, as 
that was out of LTS
by the time I discovered it.

So you could try making the above changes to channel.c and see if it improves 
the CDRs for you.

Cheers
Tony


I will try doing this change and recompiling to see what happens.

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[asterisk-users] Dial to FastAGI application appears as 1-second CDR - how do I fix?

[Alex Villací­s Lasso]

In my application, I am using AMI to run an Originate command between a channel and a dialplan application (NOT a context). In my case, the application I want to invoke is FastAGI. The Originate AMI command works correctly, but Asterisk generates a very 
short (0-1s) duration for the CDR that results from this call, regardless of the time spent running the FastAGI application. I want the CDR duration to reflect the time spent in the entirety of the call, as is normal for ordinary calls. What should I do? I 
am running Asterisk 11.25.3 .



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Re: [asterisk-users] softphone instead of desktop phones

[Alex Epshteyn]

Thomas was asking how to save money and I was just offering an option. I am 
sorry if my post was inappropriate.

That said, Thirdlane Connect itself is free, and we do offer a free version for 
companies with up to 10 users. 

-- 

Alex Epshteyn
email: a...@thirdlane.com
web: www.thirdlane.com
phone +1 415.261.6601


- Original Message -
> From: "Barry Flanagan" <barryf-li...@flanagan.ie>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> <asterisk-users@lists.digium.com>
> Sent: Sunday, April 30, 2017 11:20:25 AM
> Subject: Re: [asterisk-users] softphone instead of desktop phones
> 
> 
> 
> 
> 
> 
> On 30 April 2017 at 16:54, Tech Support < aster...@voipbusiness.us >
> wrote:
> 
> 
> 
> I thought this was a non-commercial list.
> 
> 
> 
> 
> Yeah, I wouldn't mind so much if it had actually answered the
> original poster's query. "Switch to our proprietary solution and we
> can offer you this proprietary solution" isn't a contribution, it's
> an ad.
> 
> 
> -Barry
> 
> 
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto: asterisk-users-boun...@lists.digium.com ] On Behalf Of Alex
> Epshteyn
> Sent: Saturday, April 29, 2017 08:59 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] softphone instead of desktop phones
> 
> Thirdlane Connect can be used as a softphone. It works in modern
> browsers
> (no installation is required), on Mac, Windows and Linux desktops,
> and on
> mobile phones.
> 
> Besides basic softphone functionality, it provides instant messaging,
> group
> chat (channels), voice and video conferencing, and screen sharing. It
> integrates with a variety of applications and CRMs such as
> Salesforce, Zoho,
> Zendesk, Redmine, etc.
> 
> Try it out!
> 
> 
> --
> 
> Alex Epshteyn
> web: www.thirdlane.com
> 
> 
> - Original Message -
> > From: "Amit Patkar" < a...@avhan.com >
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > < asterisk-users@lists.digium.com >
> > Sent: Saturday, April 29, 2017 9:16:05 AM
> > Subject: Re: [asterisk-users] softphone instead of desktop phones
> > 
> > 
> > Linphone is available for all major OS platforms.
> > Then there is PortGo as well
> > Regards,
> > Amit Patkar
> > 
> > 
> > On April 29, 2017 9:05:22 PM GMT+05:30, Thomas <
> > thomasit...@gmail.com >
> > wrote:
> > 
> > Hello,
> > Iam lookong for an Softphone for iPhor oder Android smartphone
> > using
> > togehter with an headset.
> > I tried Zoiper and CSipSimple but quality was bad compared to an
> > desktop SIP phone.
> > 
> > Is there an better softphone?
> > 
> > Or are there softphone solutions for PC desktop MAC or Android with
> > an
> > headset?
> > I want to save cost for desktop phones.
> > 
> > thanks Thomas
> > 
> > 
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com
> > --
> > 
> > Check out the new Asterisk community forum at:
> > https://community.asterisk.org/
> > 
> > New to Asterisk? Start here:
> > https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> > 
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
> 
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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> 
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> _
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> 
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
> 
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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> 
> 
> --
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&g

Re: [asterisk-users] softphone instead of desktop phones

[Alex Epshteyn]

Thirdlane Connect can be used as a softphone. It works in modern browsers (no 
installation is required), on Mac, Windows and Linux desktops, and on mobile 
phones.

Besides basic softphone functionality, it provides instant messaging, group 
chat (channels), voice and video conferencing, and screen sharing. It 
integrates with a variety of applications and CRMs such as Salesforce, Zoho, 
Zendesk, Redmine, etc.

Try it out!


-- 

Alex Epshteyn
web: www.thirdlane.com


- Original Message -
> From: "Amit Patkar" <a...@avhan.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> <asterisk-users@lists.digium.com>
> Sent: Saturday, April 29, 2017 9:16:05 AM
> Subject: Re: [asterisk-users] softphone instead of desktop phones
> 
> 
> Linphone is available for all major OS platforms.
> Then there is PortGo as well
> Regards,
> Amit Patkar
> 
> 
> On April 29, 2017 9:05:22 PM GMT+05:30, Thomas
> <thomasit...@gmail.com> wrote:
> 
> Hello,
> Iam lookong for an Softphone for iPhor oder Android smartphone using
> togehter
> with an headset.
> I tried Zoiper and CSipSimple but quality was bad compared to an
> desktop SIP
> phone.
> 
> Is there an better softphone?
> 
> Or are there softphone solutions for PC desktop MAC or Android with
> an
> headset?
> I want to save cost for desktop phones.
> 
> thanks Thomas
> 
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
> 
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] PBX selection

[Alex Epshteyn]

The solution you choose should be based on many factors which should include 
your business requirements, team's experience, your budget, growth expectations 
and more.

You can choose Asterisk or Freeswitch as a platform and start building on that 
- but it is not simple and being new to VoIP you are likely to make mistakes. 
The "do-it-yourself" approach will some money initially, but will be the most 
expensive option long term - as you will be denying the economy of scale. 
Bringing a "smart programmer" won't help much as you will also create a 
"lock-in". In fact, this could be worse than a dependency created when you use 
a commercial or a known open source solution as while you would still be able 
to get help from the community for the "base" part of your pbx, your custom 
part will be much harder to deal with.

Our company started building Asterisk based PBX in 2002 and Multi Tenant PBX in 
2005 - we do this as our core business and are still finding areas for 
improvement :). As your experience with VoIP is minimal I would side with your 
CTO - you should find a solution high enough in the stack to avoid the 
complexity of building it all yourself.

Good luck,

Alex

-- 

Alex Epshteyn
email: a...@thirdlane.com
web: www.thirdlane.com
phone +1 415.261.6601


- Original Message -
> From: "J Montoya or A J Stiles" <asterisk_l...@earthshod.co.uk>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> <asterisk-users@lists.digium.com>
> Sent: Tuesday, April 18, 2017 1:40:47 AM
> Subject: Re: [asterisk-users] PBX selection
> 
> On Monday 17 Apr 2017, Speed Boy wrote:
> >  Hi all, I'm new to VoIP, now we have a project that needs a
> >  PBX with client APPs.
> > In our team we have argument for choosing PBX. By so far, we
> >  have following candidates:
> > 
> > A: Open source
> > 
> >  1) Asterisk PBX (http://www.asterisk.org) (with longest
> >  history that almost every one knows it, now the last version using
> >  the
> > PJSIP stack)
> >  2) FreeSwitch (http://www.freeswitch.org) (A lot people
> >  recommended it to us)
> > 
> > 
> > B: Commercial
> > 
> > 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now
> > acquired by a HongKong company now
> > 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It
> > also includes VoIP SDK, WebRTC and offer rebranding app for free.
> > 
> > My boss prefers the Open Source PBX since they are free,
> > but our CTO prefers the commercial editions, according to
> > whom the business PBX has better support, and the
> > performance is good, and easy to use - considering our team
> > all are new to VoIP/PBX.
> 
> Proponents of proprietary solutions always like to say "If an Open
> Source
> solution breaks, who can you call?"  The answer is, "Any
> sufficiently-competent
> programmer -- it may be broken, but we have all the pieces".  Whereas
> if you
> spend money on proprietary software and it breaks, then there is only
> *one*
> place you can call -- and you'd better hope they are interested to
> fix your
> problem.
> 
> On the other hand, if you could get full Source Code and Modification
> Rights
> (basically, "everything we could do with a GPL program except
> distribute
> copies"),  a proprietary solution might not be so bad after all.  But
> since
> the goal of most proprietary software vendors is to extract money
> from you and
> maintaining you in a state of perpetual helplessness is highly
> desirable in
> the course of this, do not expect to get such a deal in real life.
>  
> > We have did some searching of Asterisk, here are my questions:
> > 
> > 1. Does the last Asterisk using PJSIP stack ?
> 
> Yes.
> 
> > 2. Does there has the comparison of PJSIP and reSIProcate,
> > sofia(using by
> > FreeSwicth) ?
> 
> Not sure about this.  We're still using the original chan_sip driver.
> 
> > 3. Is it easy to compile and setup Asterisk?
> 
> It's about as easy as compiling anything from Source Code.  Harder
> than LAME
> MP3 encoder, but easier than the Linux kernel.  If you altered
> `monop` from
> the BSDgames package to make the streets match your local edition of
> the game,
> you will have no problem whatsoever with building Asterisk.
> 
> If you understand the process of what you are doing -- basically,
> setting up
> an automated process that will examine your server hardware and
> software
> configuration  (configure),  choosing which parts of Asterisk you
> want to
> include  (make menuselect),  compiling the selected human-readable
> Sourc

[asterisk-users] Mail rejected from secur...@asterisk.org

[Alex Villacís Lasso]

: host asterisk.org.inbound10.mxlogic.net[208.65.145.3]
said: 451 Could not load DRD for domain (asterisk.org) rcpt
(secur...@asterisk.org) (in reply to RCPT TO command)

I want to inform about a potential issue but my mail to 
secur...@asterisk.org is bouncing back.
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[asterisk-users] AMI ManagerAction UpdateConfig on file with #include may corrupt the file

[Alex Villací­s Lasso]

I have seen the following scenario that may lead to a corrupted and possibly 
invalid configuration file after using UpdateConfig through AMI, at least with 
Asterisk 11.25:

There is a configuration file a.conf that contains several sections and also contains a #include "b.conf", which defines a few sections of its own. Both files a.conf and b.conf start out syntactically valid and with no duplicate categories between the two 
files.


An AMI client requests a category listing (ListCategories) from a.conf . The 
listing of categories is returned, including the ones defined in b.conf. There 
is no indication in the answer about the true source of the categories.

The AMI client attempts to use UpdateConfig with the set of actions EmptyCat, followed by a few Append to define new variables. The intent is to complete replace the content of the category with a new set of variables. It happens that the chosen category 
is actually defined in b.conf.


The actual result is that b.conf is left unmodified, and the variables that were supposed to go in the emptied category are instead prepended to the #include "b.conf" statement, without the [category] delimiter that starts a new category. This 
syntactically places the variables as a content of whatever section was before the #include, and if the #include happens to appear as the first line of the file a.conf, it leads to variables outside of any category, leading to an invalid configuration file.


Is there a way to prevent this scenario, without actually going to the filesystem and trying my hand at parsing the configuration file for any #include's that might be the true source of the categories I want to update? For the sake of argument, my AMI 
client runs on a different machine, so direct filesystem access is nontrivial.



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[asterisk-users] Representation of category templates and template inheritance in ARA Static Realtime

[Alex Villací­s Lasso]

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_UpdateConfig

According to the link above, the NewCat action allows for options that specify 
whether the newly created category is a template, and whether it should inherit 
from an existing set of templates.


https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration

The link above has a comment with an example of an ast_config table that can be 
used for ARA Static Realtime to load configuration files from my database.

Is it possible to represent templates in ARA Static Realtime? If so, how?

What will happen if I store my configuration in ARA Static Realtime and I 
attempt to create a category template using NewCat ?


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Re: [asterisk-users] Asterisk dahai install centos 7

[Alex Villacís Lasso]

El 10/12/16 a las 10:15, christopher kamutumwa escribió:
Package kernel-devel-3.10.0-327.36.3.el7.x86_64 already installed and 
latest version but i still receive the same error


[root@localhost dahdi-linux-complete-2.11.1+2.11.1]# make
make -C linux all
make[1]: Entering directory 
`/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux'

make -C drivers/dahdi/firmware firmware-loaders
make[2]: Entering directory 
`/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/firmware'
make[2]: Leaving directory 
`/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/firmware'
You do not appear to have the sources for the 3.10.0-327.el7.x86_64 
kernel installed.

make[1]: *** [modules] Error 1


When compiling kernel modules (such as DAHDI) using the system kernel 
and kernel-devel, they must be compiled under a specific kernel version. 
If no kernel version is specified, it defaults to the currently running 
kernel. In your case, the currently running kernel is 
3.10.0-327.el7.x86_64 .


For the selected kernel version, the kernel development files (supplied 
by kernel-devel) must exist under the exact same version. The installed 
kernel-devel version is 3.10.0-327.36.3.el7.x86_64 . Note carefully - 
this is NOT the same kernel version as the kernel selected for compiling -


3.10.0-327.36.3.el7.x86_64 (supplied by kernel-devel)
3.10.0-327.el7.x86_64 (selected by default)

Notice the extra ".36.3" between the two versions.

To solve this, you should do EXACTLY ONE of the following two options:
1) install the exact kernel-devel version for your currently-running 
kernel. In this case, kernel-devel-3.10.0-327.el7.x86_64 . Then run make 
as before.
2) install kernel-3.10.0-327.36.3.el7.x86_64 if not already installed, 
then specify this version as the kernel to compile DAHDI against:


make KVERS="3.10.0-327.36.3.el7.x86_64"
make install KVERS="3.10.0-327.36.3.el7.x86_64"


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[asterisk-users] Can Asterisk handle in any way an SDP with m=application webrtc-datachannel ?

[Alex Villací­s Lasso]

I have a working telephone project that uses SIP.js 0.7.5 with Asterisk on the 
server side. Currently it handles both audio and video correctly.

The SIP.js webpage has instructions for setting up a datachannel through a SIP 
call. The online demo uses OpenSIPS.

When setting up a SIP call with a datachannel through the SIP.js online demo, 
the SDP looks like this:

v=0
o=- 2849011408178506361 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS
m=application 45029 DTLS/SCTP 5000
c=IN IP4 201.234.196.170
a=candidate:1184643355 1 udp 2122260223 10.1.0.1 33877 typ host generation 0 
network-id 3
a=candidate:2680228395 1 udp 2122194687 10.0.0.245 49916 typ host generation 0 
network-id 2
a=candidate:3700057831 1 udp 2122129151 192.168.3.2 45029 typ host generation 0 
network-id 1
a=candidate:136300011 1 tcp 1518280447 10.1.0.1 9 typ host tcptype active 
generation 0 network-id 3
a=candidate:3510826715 1 tcp 1518214911 10.0.0.245 9 typ host tcptype active 
generation 0 network-id 2
a=candidate:2450102807 1 tcp 1518149375 192.168.3.2 9 typ host tcptype active 
generation 0 network-id 1
a=candidate:1976953138 1 udp 1685921535 201.234.196.170 45029 typ srflx raddr 
192.168.3.2 rport 45029 generation 0 network-id 1
a=ice-ufrag:8Dt8
a=ice-pwd:CPt/905Ehc9EKsY+Dpu4UQ7d
a=fingerprint:sha-256 
20:46:82:E4:29:DE:EB:9A:04:74:A4:31:F7:A5:7A:45:15:8B:9C:61:50:CB:8A:EE:2E:35:5A:74:10:33:09:B3
a=setup:actpass
a=mid:data
a=sctpmap:5000 webrtc-datachannel 1024
a=sendrecv
a=candidate:0dspPHO0b1Ifs6Nv 1 UDP 16777215 199.7.173.73 56338 typ relay raddr 
199.7.173.73 rport 56338

The media m= is of type application.

Is Asterisk capable of handling such a SDP, so that two SIP endpoints registered through Asterisk can begin exchanging data? From what I understand in the code, Asterisk will reject such a call. However, I want to exhaust what Asterisk can do before 
resorting to setting up a SIP proxy between the endpoint and Asterisk. I remember reading a report about a webrtc success story where the webphones were also exchanging data using datachannels.



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Re: [asterisk-users] Getting 'no shared cipher' on call to webrtc endpoint from asterisk-11.24.0

[Alex Villací­s Lasso]

El 26/10/16 a las 13:16, Alex Villací­s Lasso escribió:
I am making SIP calls using SIP.js and configuring Asterisk 11.x for websockets calls under CentOS 7. On 11.23.1 and earlier, I had to patch the code to disable auto negociation due to ASTERISK-25659. Now that the bug is supposedly fixed in commit 
8653da4fa228e1e289e09e5d024e11d24da87d94, I reverted my local patch. Calls from Firefox 49 to Asterisk connect normally. However, when Asterisk originates a call to Firefox, the call hangs up immediately with this message:


[2016-10-26 12:38:09] ERROR[2888][C-0004]: res_rtp_asterisk.c:2173 
__rtp_recvfrom: DTLS failure occurred on RTP instance '0x7f5f0802f378' due to 
reason 'no shared cipher', terminating
[2016-10-26 12:38:09] WARNING[2888][C-0004]: res_rtp_asterisk.c:3924 
ast_rtcp_read: RTCP Read error: Unspecified.  Hanging up.

I suspect this has something to do with the dtlscipher setting, which is 
currently not set anywhere in my setup.

Is my suspicion correct? How do I debug which ciphers need to be enabled?



For Asterisk 11.24.0, the elliptic curve code is #ifdef'ed around 
HAVE_OPENSSL_EC . However, nowhere in the configure.ac is a check to enable 
HAVE_OPENSSL_EC. Therefore elliptic curve is always disabled on Asterisk 
11.24.0 despite the intent of the code.




-bash-4.2# openssl ciphers -v DEFAULT
ECDHE-RSA-AES256-GCM-SHA384 TLSv1.2 Kx=ECDH Au=RSA Enc=AESGCM(256) Mac=AEAD
ECDHE-ECDSA-AES256-GCM-SHA384 TLSv1.2 Kx=ECDH Au=ECDSA Enc=AESGCM(256) 
Mac=AEAD
ECDHE-RSA-AES256-SHA384 TLSv1.2 Kx=ECDH Au=RSA  Enc=AES(256) Mac=SHA384
ECDHE-ECDSA-AES256-SHA384 TLSv1.2 Kx=ECDH Au=ECDSA Enc=AES(256)  Mac=SHA384
ECDHE-RSA-AES256-SHASSLv3 Kx=ECDH Au=RSA  Enc=AES(256) Mac=SHA1
ECDHE-ECDSA-AES256-SHA  SSLv3 Kx=ECDH Au=ECDSA Enc=AES(256) Mac=SHA1
DHE-DSS-AES256-GCM-SHA384 TLSv1.2 Kx=DH   Au=DSS Enc=AESGCM(256) Mac=AEAD
DHE-RSA-AES256-GCM-SHA384 TLSv1.2 Kx=DH   Au=RSA Enc=AESGCM(256) Mac=AEAD
DHE-RSA-AES256-SHA256   TLSv1.2 Kx=DH   Au=RSA  Enc=AES(256) Mac=SHA256
DHE-DSS-AES256-SHA256   TLSv1.2 Kx=DH   Au=DSS  Enc=AES(256) Mac=SHA256
DHE-RSA-AES256-SHA  SSLv3 Kx=DH   Au=RSA  Enc=AES(256) Mac=SHA1
DHE-DSS-AES256-SHA  SSLv3 Kx=DH   Au=DSS  Enc=AES(256) Mac=SHA1
DHE-RSA-CAMELLIA256-SHA SSLv3 Kx=DH   Au=RSA Enc=Camellia(256) Mac=SHA1
DHE-DSS-CAMELLIA256-SHA SSLv3 Kx=DH   Au=DSS Enc=Camellia(256) Mac=SHA1
ECDH-RSA-AES256-GCM-SHA384 TLSv1.2 Kx=ECDH/RSA Au=ECDH Enc=AESGCM(256) Mac=AEAD
ECDH-ECDSA-AES256-GCM-SHA384 TLSv1.2 Kx=ECDH/ECDSA Au=ECDH Enc=AESGCM(256) 
Mac=AEAD
ECDH-RSA-AES256-SHA384  TLSv1.2 Kx=ECDH/RSA Au=ECDH Enc=AES(256) Mac=SHA384
ECDH-ECDSA-AES256-SHA384 TLSv1.2 Kx=ECDH/ECDSA Au=ECDH Enc=AES(256)  Mac=SHA384
ECDH-RSA-AES256-SHA SSLv3 Kx=ECDH/RSA Au=ECDH Enc=AES(256) Mac=SHA1
ECDH-ECDSA-AES256-SHA   SSLv3 Kx=ECDH/ECDSA Au=ECDH Enc=AES(256) Mac=SHA1
AES256-GCM-SHA384   TLSv1.2 Kx=RSA  Au=RSA Enc=AESGCM(256) Mac=AEAD
AES256-SHA256   TLSv1.2 Kx=RSA  Au=RSA  Enc=AES(256) Mac=SHA256
AES256-SHA  SSLv3 Kx=RSA  Au=RSA  Enc=AES(256) Mac=SHA1
CAMELLIA256-SHA SSLv3 Kx=RSA  Au=RSA Enc=Camellia(256) Mac=SHA1
PSK-AES256-CBC-SHA  SSLv3 Kx=PSK  Au=PSK  Enc=AES(256) Mac=SHA1
ECDHE-RSA-AES128-GCM-SHA256 TLSv1.2 Kx=ECDH Au=RSA Enc=AESGCM(128) Mac=AEAD
ECDHE-ECDSA-AES128-GCM-SHA256 TLSv1.2 Kx=ECDH Au=ECDSA Enc=AESGCM(128) 
Mac=AEAD
ECDHE-RSA-AES128-SHA256 TLSv1.2 Kx=ECDH Au=RSA  Enc=AES(128) Mac=SHA256
ECDHE-ECDSA-AES128-SHA256 TLSv1.2 Kx=ECDH Au=ECDSA Enc=AES(128)  Mac=SHA256
ECDHE-RSA-AES128-SHASSLv3 Kx=ECDH Au=RSA  Enc=AES(128) Mac=SHA1
ECDHE-ECDSA-AES128-SHA  SSLv3 Kx=ECDH Au=ECDSA Enc=AES(128) Mac=SHA1
DHE-DSS-AES128-GCM-SHA256 TLSv1.2 Kx=DH   Au=DSS Enc=AESGCM(128) Mac=AEAD
DHE-RSA-AES128-GCM-SHA256 TLSv1.2 Kx=DH   Au=RSA Enc=AESGCM(128) Mac=AEAD
DHE-RSA-AES128-SHA256   TLSv1.2 Kx=DH   Au=RSA  Enc=AES(128) Mac=SHA256
DHE-DSS-AES128-SHA256   TLSv1.2 Kx=DH   Au=DSS  Enc=AES(128) Mac=SHA256
DHE-RSA-AES128-SHA  SSLv3 Kx=DH   Au=RSA  Enc=AES(128) Mac=SHA1
DHE-DSS-AES128-SHA  SSLv3 Kx=DH   Au=DSS  Enc=AES(128) Mac=SHA1
DHE-RSA-SEED-SHASSLv3 Kx=DH   Au=RSA  Enc=SEED(128) Mac=SHA1
DHE-DSS-SEED-SHASSLv3 Kx=DH   Au=DSS  Enc=SEED(128) Mac=SHA1
DHE-RSA-CAMELLIA128-SHA SSLv3 Kx=DH   Au=RSA Enc=Camellia(128) Mac=SHA1
DHE-DSS-CAMELLIA128-SHA SSLv3 Kx=DH   Au=DSS Enc=Camellia(128) Mac=SHA1
ECDH-RSA-AES128-GCM-SHA256 TLSv1.2 Kx=ECDH/RSA Au=ECDH Enc=AESGCM(128) Mac=AEAD
ECDH-ECDSA-AES128-GCM-SHA256 TLSv1.2 Kx=ECDH/ECDSA Au=ECDH Enc=AESGCM(128) 
Mac=AEAD
ECDH-RSA-AES128-SHA256  TLSv1.2 Kx=ECDH/RSA Au=ECDH Enc=AES(128) Mac=SHA256
ECDH-ECDSA-AES128-SHA256 TLSv1.2 Kx=ECDH/ECDSA Au=ECDH Enc=AES(128)  Mac=SHA256
ECDH-RSA-AES128-SHA SSLv3 Kx=ECDH/RSA Au=ECDH Enc=AES(128) Mac=SHA1
ECDH-ECDSA-AES128-SHA   SSLv3 Kx=ECDH/ECDSA Au=ECDH Enc=AES(128) Mac=SHA1
AES128-GCM-SHA256   TLSv1.2 Kx=RSA  Au=RSA Enc

[asterisk-users] Getting 'no shared cipher' on call to webrtc endpoint from asterisk-11.24.0

[Alex Villací­s Lasso]

I am making SIP calls using SIP.js and configuring Asterisk 11.x for websockets calls under CentOS 7. On 11.23.1 and earlier, I had to patch the code to disable auto negociation due to ASTERISK-25659. Now that the bug is supposedly fixed in commit 
8653da4fa228e1e289e09e5d024e11d24da87d94, I reverted my local patch. Calls from Firefox 49 to Asterisk connect normally. However, when Asterisk originates a call to Firefox, the call hangs up immediately with this message:


[2016-10-26 12:38:09] ERROR[2888][C-0004]: res_rtp_asterisk.c:2173 
__rtp_recvfrom: DTLS failure occurred on RTP instance '0x7f5f0802f378' due to 
reason 'no shared cipher', terminating
[2016-10-26 12:38:09] WARNING[2888][C-0004]: res_rtp_asterisk.c:3924 
ast_rtcp_read: RTCP Read error: Unspecified.  Hanging up.

I suspect this has something to do with the dtlscipher setting, which is 
currently not set anywhere in my setup.

Is my suspicion correct? How do I debug which ciphers need to be enabled?


-bash-4.2# openssl ciphers -v DEFAULT
ECDHE-RSA-AES256-GCM-SHA384 TLSv1.2 Kx=ECDH Au=RSA Enc=AESGCM(256) Mac=AEAD
ECDHE-ECDSA-AES256-GCM-SHA384 TLSv1.2 Kx=ECDH Au=ECDSA Enc=AESGCM(256) 
Mac=AEAD
ECDHE-RSA-AES256-SHA384 TLSv1.2 Kx=ECDH Au=RSA  Enc=AES(256) Mac=SHA384
ECDHE-ECDSA-AES256-SHA384 TLSv1.2 Kx=ECDH Au=ECDSA Enc=AES(256)  Mac=SHA384
ECDHE-RSA-AES256-SHASSLv3 Kx=ECDH Au=RSA  Enc=AES(256) Mac=SHA1
ECDHE-ECDSA-AES256-SHA  SSLv3 Kx=ECDH Au=ECDSA Enc=AES(256) Mac=SHA1
DHE-DSS-AES256-GCM-SHA384 TLSv1.2 Kx=DH   Au=DSS Enc=AESGCM(256) Mac=AEAD
DHE-RSA-AES256-GCM-SHA384 TLSv1.2 Kx=DH   Au=RSA Enc=AESGCM(256) Mac=AEAD
DHE-RSA-AES256-SHA256   TLSv1.2 Kx=DH   Au=RSA  Enc=AES(256) Mac=SHA256
DHE-DSS-AES256-SHA256   TLSv1.2 Kx=DH   Au=DSS  Enc=AES(256) Mac=SHA256
DHE-RSA-AES256-SHA  SSLv3 Kx=DH   Au=RSA  Enc=AES(256) Mac=SHA1
DHE-DSS-AES256-SHA  SSLv3 Kx=DH   Au=DSS  Enc=AES(256) Mac=SHA1
DHE-RSA-CAMELLIA256-SHA SSLv3 Kx=DH   Au=RSA Enc=Camellia(256) Mac=SHA1
DHE-DSS-CAMELLIA256-SHA SSLv3 Kx=DH   Au=DSS Enc=Camellia(256) Mac=SHA1
ECDH-RSA-AES256-GCM-SHA384 TLSv1.2 Kx=ECDH/RSA Au=ECDH Enc=AESGCM(256) Mac=AEAD
ECDH-ECDSA-AES256-GCM-SHA384 TLSv1.2 Kx=ECDH/ECDSA Au=ECDH Enc=AESGCM(256) 
Mac=AEAD
ECDH-RSA-AES256-SHA384  TLSv1.2 Kx=ECDH/RSA Au=ECDH Enc=AES(256) Mac=SHA384
ECDH-ECDSA-AES256-SHA384 TLSv1.2 Kx=ECDH/ECDSA Au=ECDH Enc=AES(256)  Mac=SHA384
ECDH-RSA-AES256-SHA SSLv3 Kx=ECDH/RSA Au=ECDH Enc=AES(256) Mac=SHA1
ECDH-ECDSA-AES256-SHA   SSLv3 Kx=ECDH/ECDSA Au=ECDH Enc=AES(256) Mac=SHA1
AES256-GCM-SHA384   TLSv1.2 Kx=RSA  Au=RSA Enc=AESGCM(256) Mac=AEAD
AES256-SHA256   TLSv1.2 Kx=RSA  Au=RSA  Enc=AES(256) Mac=SHA256
AES256-SHA  SSLv3 Kx=RSA  Au=RSA  Enc=AES(256) Mac=SHA1
CAMELLIA256-SHA SSLv3 Kx=RSA  Au=RSA Enc=Camellia(256) Mac=SHA1
PSK-AES256-CBC-SHA  SSLv3 Kx=PSK  Au=PSK  Enc=AES(256) Mac=SHA1
ECDHE-RSA-AES128-GCM-SHA256 TLSv1.2 Kx=ECDH Au=RSA Enc=AESGCM(128) Mac=AEAD
ECDHE-ECDSA-AES128-GCM-SHA256 TLSv1.2 Kx=ECDH Au=ECDSA Enc=AESGCM(128) 
Mac=AEAD
ECDHE-RSA-AES128-SHA256 TLSv1.2 Kx=ECDH Au=RSA  Enc=AES(128) Mac=SHA256
ECDHE-ECDSA-AES128-SHA256 TLSv1.2 Kx=ECDH Au=ECDSA Enc=AES(128)  Mac=SHA256
ECDHE-RSA-AES128-SHASSLv3 Kx=ECDH Au=RSA  Enc=AES(128) Mac=SHA1
ECDHE-ECDSA-AES128-SHA  SSLv3 Kx=ECDH Au=ECDSA Enc=AES(128) Mac=SHA1
DHE-DSS-AES128-GCM-SHA256 TLSv1.2 Kx=DH   Au=DSS Enc=AESGCM(128) Mac=AEAD
DHE-RSA-AES128-GCM-SHA256 TLSv1.2 Kx=DH   Au=RSA Enc=AESGCM(128) Mac=AEAD
DHE-RSA-AES128-SHA256   TLSv1.2 Kx=DH   Au=RSA  Enc=AES(128) Mac=SHA256
DHE-DSS-AES128-SHA256   TLSv1.2 Kx=DH   Au=DSS  Enc=AES(128) Mac=SHA256
DHE-RSA-AES128-SHA  SSLv3 Kx=DH   Au=RSA  Enc=AES(128) Mac=SHA1
DHE-DSS-AES128-SHA  SSLv3 Kx=DH   Au=DSS  Enc=AES(128) Mac=SHA1
DHE-RSA-SEED-SHASSLv3 Kx=DH   Au=RSA  Enc=SEED(128) Mac=SHA1
DHE-DSS-SEED-SHASSLv3 Kx=DH   Au=DSS  Enc=SEED(128) Mac=SHA1
DHE-RSA-CAMELLIA128-SHA SSLv3 Kx=DH   Au=RSA Enc=Camellia(128) Mac=SHA1
DHE-DSS-CAMELLIA128-SHA SSLv3 Kx=DH   Au=DSS Enc=Camellia(128) Mac=SHA1
ECDH-RSA-AES128-GCM-SHA256 TLSv1.2 Kx=ECDH/RSA Au=ECDH Enc=AESGCM(128) Mac=AEAD
ECDH-ECDSA-AES128-GCM-SHA256 TLSv1.2 Kx=ECDH/ECDSA Au=ECDH Enc=AESGCM(128) 
Mac=AEAD
ECDH-RSA-AES128-SHA256  TLSv1.2 Kx=ECDH/RSA Au=ECDH Enc=AES(128) Mac=SHA256
ECDH-ECDSA-AES128-SHA256 TLSv1.2 Kx=ECDH/ECDSA Au=ECDH Enc=AES(128)  Mac=SHA256
ECDH-RSA-AES128-SHA SSLv3 Kx=ECDH/RSA Au=ECDH Enc=AES(128) Mac=SHA1
ECDH-ECDSA-AES128-SHA   SSLv3 Kx=ECDH/ECDSA Au=ECDH Enc=AES(128) Mac=SHA1
AES128-GCM-SHA256   TLSv1.2 Kx=RSA  Au=RSA Enc=AESGCM(128) Mac=AEAD
AES128-SHA256   TLSv1.2 Kx=RSA  Au=RSA  Enc=AES(128) Mac=SHA256
AES128-SHA  SSLv3 Kx=RSA  Au=RSA  Enc=AES(128) Mac=SHA1
SEED-SHASSLv3 Kx=RSA  Au=RSA  Enc=SEED(128) Mac=SHA1
CAMELLIA128-SHA SSLv3 Kx=RSA  Au=RSA Enc=Camellia(128) Mac=SHA1

Re: [asterisk-users] Replacement for phpagi?

[Alex Villací­s Lasso]

El 10/08/16 a las 12:06, Carlos Chavez escribió:

Anyone know a good replacement for phpagi? Unfortunately development 
stalled long ago and it has not been updated.  What is the best solution for 
AMI and AGI on PHP? Thanks.



In the case of AMI, you could use the AMI client from the Elastix CallCenter 
dialer daemon:

https://sourceforge.net/p/elastix/code/HEAD/tree/branches/2.5.0/apps/extras/callcenter/setup/dialer_process/dialer/AMIClientConn.class.php

This class was once based on phpagi-asmanager.php but has since been completely 
rewritten to make use of an internal non-blocking I/O model. The main internal 
client is the AMIEventProcess class in the same project directory.


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Re: [asterisk-users] Trouble applying regex to dialplan variable that contains double-quotes

[Alex Villací­s Lasso]

El 08/08/16 a las 21:34, Eric Wieling escribió:


How Set handles quotes can be changed with the 'app_set' setting in the 
[compat] section of /etc/asterisk/asterisk.conf.  See also: 
https://wiki.asterisk.org/wiki/display/AST/Application_Set Perhaps you have the 
value left over from an old Asterisk setup.



I do not have any [compat] section in my /etc/asterisk/asterisk.conf file. Also the previous assignments to RX and T1 work correctly and the values are set as they appear in the dialplan. It is only the evaluation of the regex operator inside the $[ that 
gives me trouble. I thought the QUOTE() function would be the way to use a dialplan variable with special characters inside an expression, but apparently the backslash-double quote sequence is not being recognized as an escape sequence for a literal 
double-quote character in the string. See below for the actual result. So, what am I doing wrong?



On 08/08/2016 04:31 PM, Alex Villací­s Lasso wrote:
I am writing a dialplan context under asterisk 11.21.0 to handle SIP message routing between registered SIP peers using chan_sip. I am having trouble with double-quotes when the source peer uses a display name, which appears in quotes before the SIP 
URI. I want to extract the SIP URI from MESSAGE(from) in order to (conditionally) route a failure message back to the source peer.


My test dialplan sets up variables like these:

exten => _X.,n,Set(RX=".*<(.+)>")
exten => _X.,n,Set(T1="Example name" <sip:exam...@example.com>)

If I just apply the regex operator (:) on T1 using regexp RX, like this:

exten => _X.,n,Set(FROM_SIPURI=$[${T1}:${RX}])

...I get this syntax error:

[2016-08-08 15:04:02] WARNING[1653][C-]: ast_expr2.fl:470 ast_yyerror: 
ast_yyerror():  syntax error: syntax error, unexpected ':', expecting '-' or '!' or 
'(' or ''; Input:
"Example name" <sip:exam...@example.com>:".*<(.+)>"
^
(caret points at the colon character)

If I enclose the T1 variable in double quotes, like this:

exten => _X.,n,Set(FROM_SIPURI=$["${T1}":${RX}])

...I get this syntax error:

[2016-08-08 15:05:40] WARNING[1653][C-]: ast_expr2.fl:470 ast_yyerror: 
ast_yyerror():  syntax error: syntax error, unexpected '', expecting 
$end; Input:
""Example name" <sip:exam...@example.com>":".*<(.+)>"
  ^
(caret points at letter E)

If I use the QUOTE() function to quote the double quotes before applying the 
regexp, like this:

exten => _X.,n,Set(FROM_SIPURI=$[${QUOTE(${T1})}:${RX}])

... I get this syntax error:

[2016-08-08 14:53:35] WARNING[1653][C-]: ast_expr2.fl:470 ast_yyerror: 
ast_yyerror():  syntax error: syntax error, unexpected '', expecting 
$end; Input:
"\"Example name\" <sip:exam...@example.com>":".*<(.+)>"
   ^
(caret points at letter E)

Currently I am working around the issue by using REPLACE() to strip all 
double-quotes, but I believe this is not a correct solution. How should I write 
the $[ expression so that the double-quotes are handled correctly?







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[asterisk-users] Trouble applying regex to dialplan variable that contains double-quotes

[Alex Villací­s Lasso]

I am writing a dialplan context under asterisk 11.21.0 to handle SIP message routing between registered SIP peers using chan_sip. I am having trouble with double-quotes when the source peer uses a display name, which appears in quotes before the SIP URI. I 
want to extract the SIP URI from MESSAGE(from) in order to (conditionally) route a failure message back to the source peer.


My test dialplan sets up variables like these:

exten => _X.,n,Set(RX=".*<(.+)>")
exten => _X.,n,Set(T1="Example name" )

If I just apply the regex operator (:) on T1 using regexp RX, like this:

exten => _X.,n,Set(FROM_SIPURI=$[${T1}:${RX}])

...I get this syntax error:

[2016-08-08 15:04:02] WARNING[1653][C-]: ast_expr2.fl:470 ast_yyerror: 
ast_yyerror():  syntax error: syntax error, unexpected ':', expecting '-' or '!' or 
'(' or ''; Input:
"Example name" :".*<(.+)>"
^
(caret points at the colon character)

If I enclose the T1 variable in double quotes, like this:

exten => _X.,n,Set(FROM_SIPURI=$["${T1}":${RX}])

...I get this syntax error:

[2016-08-08 15:05:40] WARNING[1653][C-]: ast_expr2.fl:470 ast_yyerror: 
ast_yyerror():  syntax error: syntax error, unexpected '', expecting 
$end; Input:
""Example name" ":".*<(.+)>"
  ^
(caret points at letter E)

If I use the QUOTE() function to quote the double quotes before applying the 
regexp, like this:

exten => _X.,n,Set(FROM_SIPURI=$[${QUOTE(${T1})}:${RX}])

... I get this syntax error:

[2016-08-08 14:53:35] WARNING[1653][C-]: ast_expr2.fl:470 ast_yyerror: 
ast_yyerror():  syntax error: syntax error, unexpected '', expecting 
$end; Input:
"\"Example name\" ":".*<(.+)>"
   ^
(caret points at letter E)

Currently I am working around the issue by using REPLACE() to strip all 
double-quotes, but I believe this is not a correct solution. How should I write 
the $[ expression so that the double-quotes are handled correctly?


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[asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome

[Alex Villací­s Lasso]

I am having trouble getting Google Chrome to accept a WebRTC call coming from 
Asterisk, even though Firefox can (now) accept the same call without issue.

My setup is as follows:

Server:
CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146
asterisk-11.21.0 patched to work around 
https://issues.asterisk.org/jira/browse/ASTERISK-25659
openssl-1.0.1e-51.el7_2.2.x86_64
[root@elx4 ~]# openssl version
OpenSSL 1.0.1e-fips 11 Feb 2013
[root@elx4 ~]# openssl ecparam -list_curves
  secp384r1 : NIST/SECG curve over a 384 bit prime field
  secp521r1 : NIST/SECG curve over a 521 bit prime field
  prime256v1: X9.62/SECG curve over a 256 bit prime field

Client:
Fedora 23 x86_64
Linphone (linphone-3.6.1-10.fc23.x86_64)
Firefox 43 (firefox-43.0.3-1.fc23.x86_64)
Google Chrome (google-chrome-stable-47.0.2526.111-1.x86_64)
SIP.js 0.7.2

I set up my SIP configuration to have two SIP accounts. Account 1000 is the 
Linphone and 1001 is the webrtc:

[general]
faxdetect=no
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-2.11.0(11.20.0)
disallow=all
allow=g723
allow=ulaw
allow=gsm
allow=alaw
allow=g729
allow=speex
allow=g722
allow=h264
allow=h263p
allow=h263
allow=h261
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscipher=ALL
tlsclientmethod=tlsv1
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
callevents=no
jbenable=no
videosupport=yes
allowguest=no
srvlookup=no
defaultexpiry=120
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=20
g726nonstandard=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
checkmwi=10
notifyringing=yes
notifyhold=yes
nat=yes

[1000]
deny=0.0.0.0/0.0.0.0
secret=6ff108122cce3b0b45e0abf374c14ef4
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
icesupport=no
dtlsenable=no
dtlsverify=no
dtlssetup=actpass
encryption=no
callgroup=
pickupgroup=
dial=SIP/1000
mailbox=1000@device
permit=0.0.0.0/0.0.0.0
callerid=Usuario 1 elx4 <1000>
callcounter=yes
faxdetect=no

[1001]
deny=0.0.0.0/0.0.0.0
secret=ce93963b0751ed9a88ec1badbc073fce
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=wss,ws,udp,tcp,tls
avpf=yes
icesupport=yes
dtlsenable=yes
dtlsverify=no
dtlssetup=actpass
dtlscertfile=/var/lib/asterisk/keys/localhost.crt
dtlsprivatekey=/var/lib/asterisk/keys/localhost.key
encryption=yes
callgroup=
pickupgroup=
dial=SIP/1001
mailbox=1001@device
permit=0.0.0.0/0.0.0.0
callerid=Usuario Alex <1001>
callcounter=yes
faxdetect=no


With this setup, I can make calls using the SIP softphone as usual, both into and out of asterisk. After approving the certificate exceptions, I can also use the webrtc account to generate a call from either Firefox or Google Chrome into asterisk, out to 
the SIP softphone.


The problem arises when I try to make asterisk send a call into the browser. When using Firefox 43 I can receive the call normally (this required patching around ASTERISK-25659) and all is well. However, in Google Chrome, the call is rejected with a 
message of "Failed to set remote video description send parameters.." as shown in this SIP trace in the browser console:




Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.transport | received WebSocket 
text message:

INVITE sip:8dgpkoa2@192.0.2.210;transport=wss SIP/2.0
Via: SIP/2.0/WS 10.1.0.4:5060;branch=z9hG4bK4f80b96d;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as37a33245
To: <sip:8dgpkoa2@192.0.2.210;transport=wss>
Contact: <sip:anonymous@10.1.0.4:5060;transport=WS>
Call-ID: 61c6be5b44d587c80b56f98e756037ab@10.1.0.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.20.0)
Date: Wed, 20 Jan 2016 18:54:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1799

v=0
o=root 469858785 469858785 IN IP4 10.1.0.4
s=Asterisk PBX 11.21.0
c=IN IP4 10.1.0.4
b=CT:384
t=0 0
m=audio 14814 UDP/TLS/RTP/SAVPF 4 0 3 8 18 110 9 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:0e746cd50c88ce6e383ff3882acebb80
a=ice-pwd:1a9a09862254ae253f06a0bb184fd1b5
a=candidate:Ha010004 1 UDP 2130706431 10.1.0.4 14814 typ host
a=candidate:Hc0a80592 1 UDP 2130706431 192.168.5.146 14814 typ host
a=candidate:Ha010004 2 UDP 2130706430 10.1.0.4 14815 typ host
a=candidate:Hc0a80592 2 UDP 2130706430 192.168.5.146 14815 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 

Re: [asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome

[Alex Villací­s Lasso]

El 20/01/16 a las 16:25, Alex Villací­s Lasso escribió:

I am having trouble getting Google Chrome to accept a WebRTC call coming from 
Asterisk, even though Firefox can (now) accept the same call without issue.

My setup is as follows:

Server:
CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146
asterisk-11.21.0 patched to work around 
https://issues.asterisk.org/jira/browse/ASTERISK-25659
openssl-1.0.1e-51.el7_2.2.x86_64
[root@elx4 ~]# openssl version
OpenSSL 1.0.1e-fips 11 Feb 2013
[root@elx4 ~]# openssl ecparam -list_curves
  secp384r1 : NIST/SECG curve over a 384 bit prime field
  secp521r1 : NIST/SECG curve over a 521 bit prime field
  prime256v1: X9.62/SECG curve over a 256 bit prime field

Client:
Fedora 23 x86_64
Linphone (linphone-3.6.1-10.fc23.x86_64)
Firefox 43 (firefox-43.0.3-1.fc23.x86_64)
Google Chrome (google-chrome-stable-47.0.2526.111-1.x86_64)
SIP.js 0.7.2

I set up my SIP configuration to have two SIP accounts. Account 1000 is the 
Linphone and 1001 is the webrtc:

[general]
faxdetect=no
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-2.11.0(11.20.0)
disallow=all
allow=g723
allow=ulaw
allow=gsm
allow=alaw
allow=g729
allow=speex
allow=g722
allow=h264
allow=h263p
allow=h263
allow=h261
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscipher=ALL
tlsclientmethod=tlsv1
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
callevents=no
jbenable=no
videosupport=yes
allowguest=no
srvlookup=no
defaultexpiry=120
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=20
g726nonstandard=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
checkmwi=10
notifyringing=yes
notifyhold=yes
nat=yes

[1000]
deny=0.0.0.0/0.0.0.0
secret=6ff108122cce3b0b45e0abf374c14ef4
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
icesupport=no
dtlsenable=no
dtlsverify=no
dtlssetup=actpass
encryption=no
callgroup=
pickupgroup=
dial=SIP/1000
mailbox=1000@device
permit=0.0.0.0/0.0.0.0
callerid=Usuario 1 elx4 <1000>
callcounter=yes
faxdetect=no

[1001]
deny=0.0.0.0/0.0.0.0
secret=ce93963b0751ed9a88ec1badbc073fce
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=wss,ws,udp,tcp,tls
avpf=yes
icesupport=yes
dtlsenable=yes
dtlsverify=no
dtlssetup=actpass
dtlscertfile=/var/lib/asterisk/keys/localhost.crt
dtlsprivatekey=/var/lib/asterisk/keys/localhost.key
encryption=yes
callgroup=
pickupgroup=
dial=SIP/1001
mailbox=1001@device
permit=0.0.0.0/0.0.0.0
callerid=Usuario Alex <1001>
callcounter=yes
faxdetect=no


With this setup, I can make calls using the SIP softphone as usual, both into and out of asterisk. After approving the certificate exceptions, I can also use the webrtc account to generate a call from either Firefox or Google Chrome into asterisk, out to 
the SIP softphone.


The problem arises when I try to make asterisk send a call into the browser. When using Firefox 43 I can receive the call normally (this required patching around ASTERISK-25659) and all is well. However, in Google Chrome, the call is rejected with a 
message of "Failed to set remote video description send parameters.." as shown in this SIP trace in the browser console:




Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.transport | received WebSocket 
text message:

INVITE sip:8dgpkoa2@192.0.2.210;transport=wss SIP/2.0
Via: SIP/2.0/WS 10.1.0.4:5060;branch=z9hG4bK4f80b96d;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as37a33245
To: <sip:8dgpkoa2@192.0.2.210;transport=wss>
Contact: <sip:anonymous@10.1.0.4:5060;transport=WS>
Call-ID: 61c6be5b44d587c80b56f98e756037ab@10.1.0.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.20.0)
Date: Wed, 20 Jan 2016 18:54:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1799

v=0
o=root 469858785 469858785 IN IP4 10.1.0.4
s=Asterisk PBX 11.21.0
c=IN IP4 10.1.0.4
b=CT:384
t=0 0
m=audio 14814 UDP/TLS/RTP/SAVPF 4 0 3 8 18 110 9 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:0e746cd50c88ce6e383ff3882acebb80
a=ice-pwd:1a9a09862254ae253f06a0bb184fd1b5
a=candidate:Ha010004 1 UDP 2130706431 10.1.0.4 14814 typ host
a=candidate:Hc0a80592 1 UDP 2130706431 192.168.5.146 14814 typ host
a=candidate:Ha010004 2 UDP 2130706430 10.1.0.4 14815 typ host
a=candidate:Hc0a80592 2 UDP 2130706430 192.168.5.146 14815 typ host

Re: [asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome

[Alex Villací­s Lasso]

El 20/01/16 a las 18:33, Alex Villací­s Lasso escribió:

El 20/01/16 a las 16:25, Alex Villací­s Lasso escribió:

Partial fix: Google Chrome accepts the call if videosupport is set to "no". 
This is the SDP of the successful INVITE that Chrome accepts:

INVITE sip:8cj802p8@192.0.2.240;transport=wss SIP/2.0
Via: SIP/2.0/WS 10.1.0.4:5060;branch=z9hG4bK65071dc5;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as474012b5
To: <sip:8cj802p8@192.0.2.240;transport=wss>
Contact: <sip:anonymous@10.1.0.4:5060;transport=WS>
Call-ID: 73b82a5b6fbaab50741cd99424b1f31a@10.1.0.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.20.0)
Date: Wed, 20 Jan 2016 23:27:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 937

v=0
o=root 2094440180 2094440180 IN IP4 10.1.0.4
s=Asterisk PBX 11.21.0
c=IN IP4 10.1.0.4
t=0 0
m=audio 18758 UDP/TLS/RTP/SAVPF 4 0 3 8 18 110 9 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:0033e2a20fe4becd1c34b13f5efcf1e3
a=ice-pwd:65693b30588f061710baa3584253eaba
a=candidate:Ha010004 1 UDP 2130706431 10.1.0.4 18758 typ host
a=candidate:Hc0a80592 1 UDP 2130706431 192.168.5.146 18758 typ host
a=candidate:Ha010004 2 UDP 2130706430 10.1.0.4 18759 typ host
a=candidate:Hc0a80592 2 UDP 2130706430 192.168.5.146 18759 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 
A1:9E:D0:11:26:AA:30:00:AF:06:87:9C:A7:C2:70:4F:A3:3F:89:B5:7D:5C:FA:90:89:7E:D0:8A:F0:72:F5:2A
a=sendrecv

However, I want to enable full video passthrough. Is this some kind of video 
codec incompatibility?



If I enable allow=vp8 to the set of allowed codecs, Chrome accepts the video 
call, but now I get no sound with the demo-congrats command.

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[asterisk-users] We're one week away from OpenSIPS Week in Austin!

[Alex Goulis]

We're almost there everyone!

The OpenSIPS Summit in Austin is 1 week away. There's still some room 
left for all you procrastinators!


We also have a few spots left for the OpenSIPS LIVE Bootcamp following 
the summit. Enjoy a rare opportunity to train with  the founder of the 
OpenSIPS Project Bogdan-Andrei Iancu!


Help us "Keep Austin Wierd" by celebrating OpenSIPS Week in Austin, TX 
and become an OpenSIPS Certified Professional in the process!


As part of OpenSIPS week in Austin, we'll be holding a LIVE training 
following the conclusion of the OpenSIPS Summit on Nov 10th. If you've 
ever wanted to become an OpenSIPS Certified Professional, this is you're 
chance! You'll have the rare opportunity to train live with the founder 
of the OpenSIPS Project Bogdan-Andrei Iancu! It's not a secret that 
there is a growing demand for OpenSIPS Certified Professionals all over 
the world. This training is intended to offer you a 50/50 view between 
theoretical and practical applications of OpenSIPS. It covers advanced 
topics that are critical to telecom network operators in regards to 
security, scaling, and integration. For participants that would like an 
introduction to OpenSIPS basics and to qualify for prerequisites to the 
advanced topics, there will be a 2 day online primer prior to the live 
training.


On November 11th -13th 2015 the live portion of the training will be 
held at the OMNI Downtown Austin in Austin, TX. This is our preferred 
hotel for the summit. If you're attending the summit and staying there, 
you'll enjoy the convenience of not having to leave the hotel. Prior to 
the live training, we'll be offering the first 2 days of the training 
online.


To register and get more information go to: http://ebootcamp.opensips.org

OpenSIPS Summit in Austin November 9th-10th
To register and get more information go to: 
http://www.opensips.org/Community/Summit-2015Austin


HOTEL
Hotel information can be found at: 
http://www.opensips.org/Community/Summit-2015Austin#toc5


TRAVEL INFORMATION
Airport: Austin=Bergstrom International Airport
Website: http://www.austintexas.gov/airport

This "can't miss" training is coming to you via a joint effort between 
OpenSIPS Solutions and V.Office Networks. Don't miss this rare 
opportunity to train live with the founder of the OpenSIPS Project and 
become part of the growing group of OpenSIPS Certified Professionals all 
over the world!





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[asterisk-users] Detect queue agent in wrapuptime period through AMI

[Alex Villací­s Lasso]

Is there a way to use AMI to detect whether an agent that appears to be free is 
in its wrap-up-time period?

I am using AMI to query the queue status and its members, in order to generate 
calls directed to the queue, and I do not want to originate calls if some of 
them will not be assigned immediately to agents because of the wrap-up time.

As far as I can see in the Asterisk 11 source code, the QueueStatus AMI command 
does not report the wrapuptime value, and the QueueMember event does not report 
whether the agent is waiting its wrapup time either.

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[asterisk-users] tls forasterisk

[Alex Vlad]

hi, 
is there some tutorials on the you tube how to setup tls for asterisk and how 
to get ip phones register like thatwhat ip phones can support tls any 
way?Alex
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[asterisk-users] Why is CDR(recordingfile) not being written to the database despite being set in the dialplan?

[Alex Villací­s Lasso]

I am using Asterisk 11.17.1 with my program that uses AMI Originate calls to generate a bunch of calls for a callcenter. The PBX configuration is handled by FreePBX 2.11. I want to understand the dialplan behavior in order to figure out why the 
CDR(recordingfile) is blank on the CDR records despite the dialplan setting it.


My program generates the calls by setting Channel=Local/NUMBERTODIAL@from-internal and by setting Exten=QUEUENUM, Context=from-internal, Priority=1 . The FreePBX-generated dialplan results in the following trace as seen in the Asterisk console. In 
particular, please note that CDR(recordingfile) is being set in Local/9991416445@from-internal-017b;1 :


-- Executing [9991416445@from-internal:1] 
Macro(Local/9991416445@from-internal-017b;2, 
user-callerid,LIMIT,EXTERNAL,) in new stack
-- Executing [s@macro-user-callerid:1] 
Set(Local/9991416445@from-internal-017b;2, 
TOUCH_MONITOR=1429224932.21697) in new stack
-- Executing [s@macro-user-callerid:2] 
Set(Local/9991416445@from-internal-017b;2, AMPUSER=9991416445) in new 
stack
-- Executing [s@macro-user-callerid:3] 
GotoIf(Local/9991416445@from-internal-017b;2, 0?report) in new stack
...
-- Executing [s@sub-record-check:19] Set(Local/9991416445@from-internal-017b;2, 
__TIMESTR=20150416-175532) in new stack
-- Executing [s@sub-record-check:20] Set(Local/9991416445@from-internal-017b;2, 
__FROMEXTEN=9991416445) in new stack
-- Executing [s@sub-record-check:21] Set(Local/9991416445@from-internal-017b;2, 
__CALLFILENAME=out-9991416445-9991416445-20150416-175532-1429224932.21697) in new stack
-- Executing [s@sub-record-check:22] 
Goto(Local/9991416445@from-internal-017b;2, out,1) in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] 
ExecIf(Local/9991416445@from-internal-017b;2, 
1?Set(__REC_POLICY_MODE=)) in new stack
-- Executing [out@sub-record-check:2] 
GosubIf(Local/9991416445@from-internal-017b;2, 
0?record,1(exten,9991416445,9991416445)) in new stack
-- Executing [out@sub-record-check:3] 
Return(Local/9991416445@from-internal-017b;2, ) in new stack
-- Executing [9991416445@from-internal:5] 
PlayTones(Local/9991416445@from-internal-017b;2, ring) in new stack
-- Executing [9991416445@from-internal:6] 
Dial(Local/9991416445@from-internal-017b;2, 
SIP/5547741200/019991416445,40,tTorR) in new stack
...
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/5547741200/0459991416445
-- SIP/5547741200-51cb is ringing

-- SIP/5547741200-51cb is ringing
-- SIP/5547741200-51cb is making progress passing it to 
Local/9991416445@from-internal-017b;2

-- SIP/5547741200-51cb is making progress passing it to 
Local/9991416445@from-internal-017b;2

-- SIP/5547741200-51cb is making progress passing it to 
Local/9991416445@from-internal-017b;2

-- SIP/5547741200-51cb is ringing
-- SIP/5547741200-51cb is making progress passing it to 
Local/9991416445@from-internal-017b;2
-- SIP/5547741200-51cb answered 
Local/9991416445@from-internal-017b;2
-- Executing [6001@from-internal:1] 
Macro(Local/9991416445@from-internal-017b;1, user-callerid,) in new 
stack
-- Executing [s@macro-user-callerid:1] 
Set(Local/9991416445@from-internal-017b;1, 
TOUCH_MONITOR=1429224932.21696) in new stack
-- Executing [s@macro-user-callerid:2] 
Set(Local/9991416445@from-internal-017b;1, AMPUSER=9991416445) in new 
stack
-- Executing [s@macro-user-callerid:3] 
GotoIf(Local/9991416445@from-internal-017b;1, 0?report) in new stack
-- Executing [s@macro-user-callerid:4] 
ExecIf(Local/9991416445@from-internal-017b;1, 
1?Set(REALCALLERIDNUM=9991416445)) in new stack
-- Executing [s@macro-user-callerid:5] 
Set(Local/9991416445@from-internal-017b;1, AMPUSER=) in new stack
...
-- Executing [6001@from-internal:29] Set(Local/9991416445@from-internal-017b;1, 
VQ_POSITION=) in new stack
-- Executing [6001@from-internal:30] Set(Local/9991416445@from-internal-017b;1, 
__MIXMON_FORMAT=wav) in new stack
-- Executing [6001@from-internal:31] Set(Local/9991416445@from-internal-017b;1, 
MONITOR_OPTIONS=b) in new stack
-- Executing [6001@from-internal:32] 
Gosub(Local/9991416445@from-internal-017b;1, 
sub-record-check,s,1(q,6001,always)) in new stack
-- Executing [s@sub-record-check:1] Set(Local/9991416445@from-internal-017b;1, 
REC_POLICY_MODE_SAVE=) in new stack
-- Executing [s@sub-record-check:2] 
GotoIf(Local/9991416445@from-internal-017b;1, 1?check) in new stack
-- Goto (sub-record-check,s,7)
-- Executing [s@sub-record-check:7] Set(Local/9991416445@from-internal-017b;1, 
__MON_FMT=wav) in new stack
...
-- Executing [q@sub-record-check:1] 
GosubIf(Local/9991416445@from-internal-017b;1, 
1?recq,1(q,6001,9991416445)) in new stack
-- 

[asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

[Alex Villací­s Lasso]

I am trying to collect enough information about an problem a client is having 
with its asterisk 11.17.0  x86_64. This issue was observed before in 1.8.20, 
and we upgraded to 11.15.0 and then to 11.17.0 with no solution.

Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses asterisk with FreePBX 2.8. In order to generate the calls, I wrote a program that 
connects to Asterisk using the AMI protocol. This program expects the SIP agent extensions to be assigned as members of queues, of which there are about 20, as shown below:


9007 has 0 calls (max unlimited) in 'random' strategy (5s holdtime, 68s 
talktime), W:0, C:581, A:260, SL:82.6% within 60s
   Members:
  SIP/147 (ringinuse disabled) (dynamic) (On Hold) has taken 21 calls (last 
was 800 secs ago)
  SIP/417 (ringinuse disabled) (dynamic) (In use) has taken 77 calls (last 
was 708 secs ago)
  SIP/416 (ringinuse disabled) (dynamic) (In use) has taken 41 calls (last 
was 656 secs ago)
  SIP/408 (ringinuse disabled) (dynamic) (In use) has taken 50 calls (last 
was 789 secs ago)
   No Callers

The program runs queue show through AMI every few seconds. For each queue to be used in telemarketing, the program counts the number of members that are Not In Use. If at least one is found, it reads that many phone numbers from the database and uses 
the AMI Originate command on each one, as follows:


Action: Originate
Channel: Local/NN@from-internal
Exten: 
Context: from-internal
Priority: 1
Async: true
ActionID: xxx

Here, NN is the number read from the database and  is the queue extension in the FreePBX-created context that eventually runs the Queue() dialplan application for the corresponding queue. This causes the call to be connected between the 
outgoing number and the queue, and is then assigned to a queue member by Asterisk. The dialplan is configured to route NN through one of a series of SIP trunks using the outbound routes as configured by FreePBX.


The issue is that although this strategy works correctly on the user's machine for a few days, we have been observing that eventually the application stops placing calls. The agents are all idle (all 90 to 100 of them), but the queue show command shows 
them to be In Use on all queues. Furthermore, in normal operation, the core show channels command shows at most one channel for each configured SIP client in the Up state, but when calls stop being placed, the same command reports multiple channels 
in the Up state, as follows (after sorting):


Local/9757007441@from-internal-a447;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5557007441,300,!47740412!!!3!572!(None)!1428426012.169192
Local/9759315789@from-internal-a456;1ZOMBIE!from-trunk-sip-5547740412!!1!Up!AppDial!(Outgoing
 Line)!9759315789!!!3!500!(None)!1428426084.169326
Local/9759315789@from-internal-a456;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5559315789,300,!47740412!!!3!500!(None)!1428426084.169323
SIP/104-00014c61!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/0453511309468,300,!47740413!!!3!562!SIP/5547740413-00014c62!1428426022.169224
SIP/110-00014c2b!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing 
Line)!110!!!3!590!(None)!1428425994.169124
SIP/110-00014e4e!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552114757,300,!47740413!!!3!92!(None)!1428426491.169760
SIP/113-00014c8c!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740420/0016499465293,300,!47740420!!!3!532!(None)!1428426052.169273
SIP/114-00014ce6!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/040,300,!47740410!!!3!430!(None)!1428426154.169384
SIP/115-00014ea4!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740400/0059144681666,300,!47740400!!!3!10!(None)!1428426574.169850
SIP/119-00014c26!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/016255863252,300,!47740413!!!3!593!(None)!1428425991.169113
SIP/119-00014d1a!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552716011,300,!47740413!!!3!383!(None)!1428426201.169436
SIP/119-00014d4d!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5556802622,300,!47740413!!!3!327!(None)!1428426257.169493
SIP/120-00014d5e!macro-dial-one!s!37!Up!Dial!SIP/230,,trT!120!!!3!314!SIP/230-00014d5f!1428426270.169510
SIP/121-00014c24!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing 
Line)!121!!!3!596!(None)!1428425988.169111
SIP/122-00014b56!EjecutivoQUADS!93000!1!Up!AppQueue!(Outgoing 
Line)!122!!!3!677!(None)!1428425906.168693
SIP/123-00014d53!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/017222497260,300,!47740410!!!3!320!(None)!1428426264.169499
SIP/123-00014e35!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/017222497260,300,!47740410!!!3!121!(None)!1428426463.169735
SIP/123-00014e9e!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740410/5556350254,300,!47740410!!!3!13!(None)!1428426570.169844

Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

[Alex Villací­s Lasso]

El 07/04/15 a las 17:38, Alex Villací­s Lasso escribió:

I am trying to collect enough information about an problem a client is having 
with its asterisk 11.17.0  x86_64. This issue was observed before in 1.8.20, 
and we upgraded to 11.15.0 and then to 11.17.0 with no solution.

Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses asterisk with FreePBX 2.8. In order to generate the calls, I wrote a program that 
connects to Asterisk using the AMI protocol. This program expects the SIP agent extensions to be assigned as members of queues, of which there are about 20, as shown below:


9007 has 0 calls (max unlimited) in 'random' strategy (5s holdtime, 68s 
talktime), W:0, C:581, A:260, SL:82.6% within 60s
   Members:
  SIP/147 (ringinuse disabled) (dynamic) (On Hold) has taken 21 calls (last 
was 800 secs ago)
  SIP/417 (ringinuse disabled) (dynamic) (In use) has taken 77 calls (last 
was 708 secs ago)
  SIP/416 (ringinuse disabled) (dynamic) (In use) has taken 41 calls (last 
was 656 secs ago)
  SIP/408 (ringinuse disabled) (dynamic) (In use) has taken 50 calls (last 
was 789 secs ago)
   No Callers

The program runs queue show through AMI every few seconds. For each queue to be used in telemarketing, the program counts the number of members that are Not In Use. If at least one is found, it reads that many phone numbers from the database and uses 
the AMI Originate command on each one, as follows:


Action: Originate
Channel: Local/NN@from-internal
Exten: 
Context: from-internal
Priority: 1
Async: true
ActionID: xxx

Here, NN is the number read from the database and  is the queue extension in the FreePBX-created context that eventually runs the Queue() dialplan application for the corresponding queue. This causes the call to be connected between the 
outgoing number and the queue, and is then assigned to a queue member by Asterisk. The dialplan is configured to route NN through one of a series of SIP trunks using the outbound routes as configured by FreePBX.


The issue is that although this strategy works correctly on the user's machine for a few days, we have been observing that eventually the application stops placing calls. The agents are all idle (all 90 to 100 of them), but the queue show command shows 
them to be In Use on all queues. Furthermore, in normal operation, the core show channels command shows at most one channel for each configured SIP client in the Up state, but when calls stop being placed, the same command reports multiple channels 
in the Up state, as follows (after sorting):


Local/9757007441@from-internal-a447;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5557007441,300,!47740412!!!3!572!(None)!1428426012.169192
Local/9759315789@from-internal-a456;1ZOMBIE!from-trunk-sip-5547740412!!1!Up!AppDial!(Outgoing
 Line)!9759315789!!!3!500!(None)!1428426084.169326
Local/9759315789@from-internal-a456;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5559315789,300,!47740412!!!3!500!(None)!1428426084.169323
SIP/104-00014c61!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/0453511309468,300,!47740413!!!3!562!SIP/5547740413-00014c62!1428426022.169224
SIP/110-00014c2b!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing 
Line)!110!!!3!590!(None)!1428425994.169124
SIP/110-00014e4e!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552114757,300,!47740413!!!3!92!(None)!1428426491.169760
SIP/113-00014c8c!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740420/0016499465293,300,!47740420!!!3!532!(None)!1428426052.169273
SIP/114-00014ce6!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/040,300,!47740410!!!3!430!(None)!1428426154.169384
SIP/115-00014ea4!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740400/0059144681666,300,!47740400!!!3!10!(None)!1428426574.169850
SIP/119-00014c26!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/016255863252,300,!47740413!!!3!593!(None)!1428425991.169113
SIP/119-00014d1a!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552716011,300,!47740413!!!3!383!(None)!1428426201.169436
SIP/119-00014d4d!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5556802622,300,!47740413!!!3!327!(None)!1428426257.169493
SIP/120-00014d5e!macro-dial-one!s!37!Up!Dial!SIP/230,,trT!120!!!3!314!SIP/230-00014d5f!1428426270.169510
SIP/121-00014c24!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing 
Line)!121!!!3!596!(None)!1428425988.169111
SIP/122-00014b56!EjecutivoQUADS!93000!1!Up!AppQueue!(Outgoing 
Line)!122!!!3!677!(None)!1428425906.168693
SIP/123-00014d53!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/017222497260,300,!47740410!!!3!320!(None)!1428426264.169499
SIP/123-00014e35!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/017222497260,300,!47740410!!!3!121!(None)!1428426463.169735
SIP/123-00014e9e!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740410/5556350254,300,!47740410!!!3!13!(None)!1428426570.169844
SIP/125-00014bc7!macro-dialout

Re: [asterisk-users] Showing sip subscriptions in Manager

[Alex Epshteyn]

You can use Command command, and sip show subscriptions as a parameter

-- 

Alex Epshteyn
email: a...@thirdlane.com
web: www.thirdlane.com
phone +1 415.261.6601


- Original Message -
 From: Leandro Dardini ldard...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Thursday, January 15, 2015 3:00:30 PM
 Subject: [asterisk-users] Showing sip subscriptions in Manager
 
 
 
 Hello,
 almost any useful CLI command has an analogue on Asterisk Manager
 Interface, but I cannot find a way to get the list of subscriptions
 using AMI. Which is the command, if any? The CLI command is sip
 show subscriptions
 
 
 Leandro
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[asterisk-users] Problems linking asterisk against self-compiled openssl on CentOS 5

[Alex Villací­s Lasso]

I am trying to enable full WebRTC support on asterisk-11.15 for installation on a CentOS 5 machine. Currently the distro cannot be upgraded to any later CentOS series. This CentOS series ships with openssl-0.9.8e, which lacks DTLS-SRTP support required for 
WebRTC. So I decided to build a parallel install of openssl. I chose the Fedora 21 package, openssl-1.0.1j, and built it on CentOS 5. The libraries now reside at /opt/openssl101/usr/lib with header files at /opt/openssl101/usr/include/openssl/ . There are 
symbolic links at /usr/lib64/libssl.so.10 and /usr/lib64/libcrypto.so.10 pointing into my upgraded openssl.


Now I am trying to compile asterisk and link it with my openssl. My configure 
invocation is as follows:

./configure --build=x86_64-redhat-linux-gnu --host=x86_64-redhat-linux-gnu --target=x86_64-redhat-linux-gnu --program-prefix= --prefix=/usr --exec-prefix=/usr --bindir=/usr/bin --sbindir=/usr/sbin --sysconfdir=/etc --datadir=/usr/share 
--includedir=/usr/include --libdir=/usr/lib64 --libexecdir=/usr/libexec --localstatedir=/var --sharedstatedir=/usr/com --mandir=/usr/share/man --infodir=/usr/share/info --with-misdn --with-sounds-cache=no --with-srtp --with-ssl=/opt/openssl101/usr 
--with-crypto=/opt/openssl101/usr


Note the --with-ssl and --with-crypto options at the end, pointing to my 
openssl directory.

After this I compile, but I am getting these messages when compilation reaches 
res/res_rtp_asterisk.c:

a - output/pjlib-x86_64-redhat-linux-gnu/sock_qos_bsd.o
a - output/pjlib-x86_64-redhat-linux-gnu/ssl_sock_common.o
a - output/pjlib-x86_64-redhat-linux-gnu/ssl_sock_ossl.o
a - output/pjlib-x86_64-redhat-linux-gnu/ssl_sock_dump.o
a - output/pjlib-x86_64-redhat-linux-gnu/string.o
a - output/pjlib-x86_64-redhat-linux-gnu/timer.o
a - output/pjlib-x86_64-redhat-linux-gnu/types.o
   [CC] res_rtp_asterisk.c - res_rtp_asterisk.o
res_rtp_asterisk.c: In function 'ast_rtp_dtls_set_configuration':
res_rtp_asterisk.c:1278: warning: implicit declaration of function 
'SSL_CTX_set_tlsext_use_srtp'
res_rtp_asterisk.c: In function 'dtls_srtp_handle_timeout':
res_rtp_asterisk.c:1765: warning: implicit declaration of function 
'DTLSv1_handle_timeout'
res_rtp_asterisk.c: In function 'dtls_srtp_check_pending':
res_rtp_asterisk.c:1817: warning: implicit declaration of function 
'DTLSv1_get_timeout'
res_rtp_asterisk.c: In function 'dtls_srtp_setup':
res_rtp_asterisk.c:1904: warning: implicit declaration of function 
'SSL_export_keying_material'
   [LD] res_rtp_asterisk.o - res_rtp_asterisk.so
   [CC] res_rtp_multicast.c - res_rtp_multicast.o
   [LD] res_rtp_multicast.o - res_rtp_multicast.so

After this, res_rtp_asterisk.so cannot be loaded because it is linked to the 
system openssl but requests the symbols from the upgraded openssl:

[2014-12-22 20:19:41] WARNING[25901] loader.c: Error loading module 
'res_rtp_asterisk.so': /usr/lib64/asterisk/modules/res_rtp_asterisk.so: 
undefined symbol: DTLSv1_handle_timeout

[root@rpmbuild64-2 ~]# ldd /usr/lib64/asterisk/modules/res_rtp_asterisk.so
linux-vdso.so.1 =  (0x7fff431fd000)
libuuid.so.1 = /lib64/libuuid.so.1 (0x2b3f92461000)
libm.so.6 = /lib64/libm.so.6 (0x2b3f92665000)
libnsl.so.1 = /lib64/libnsl.so.1 (0x2b3f928e9000)
librt.so.1 = /lib64/librt.so.1 (0x2b3f92b01000)
libpthread.so.0 = /lib64/libpthread.so.0 (0x2b3f92d0a000)
libcrypto.so.6 = /lib64/libcrypto.so.6 (0x2b3f92f27000)
libssl.so.6 = /lib64/libssl.so.6 (0x2b3f93278000)
libc.so.6 = /lib64/libc.so.6 (0x2b3f934c6000)
/lib64/ld-linux-x86-64.so.2 (0x0037c240)
libdl.so.2 = /lib64/libdl.so.2 (0x2b3f9382)
libz.so.1 = /lib64/libz.so.1 (0x2b3f93a24000)
libgssapi_krb5.so.2 = /usr/lib64/libgssapi_krb5.so.2 (0x2b3f93c38000)
libkrb5.so.3 = /usr/lib64/libkrb5.so.3 (0x2b3f93e67000)
libcom_err.so.2 = /lib64/libcom_err.so.2 (0x2b3f940fc000)
libk5crypto.so.3 = /usr/lib64/libk5crypto.so.3 (0x2b3f942fe000)
libkrb5support.so.0 = /usr/lib64/libkrb5support.so.0 (0x2b3f94524000)
libkeyutils.so.1 = /lib64/libkeyutils.so.1 (0x2b3f9472c000)
libresolv.so.2 = /lib64/libresolv.so.2 (0x2b3f9492e000)
libselinux.so.1 = /lib64/libselinux.so.1 (0x2b3f94b44000)
libsepol.so.1 = /lib64/libsepol.so.1 (0x2b3f94d5c000)
(using libcrypto.so.6 and libssl.so.6 from system openssl)

However, libasteriskssl.so is correctly linked against my upgraded openssl:

[root@rpmbuild64-2 ~]# ldd /usr/lib64/libasteriskssl.so.1
linux-vdso.so.1 =  (0x7fffac39)
libssl.so.10 = /usr/lib64/libssl.so.10 (0x2aef0cfb4000)
libcrypto.so.10 = /usr/lib64/libcrypto.so.10 (0x2aef0d21d000)
libpthread.so.0 = /lib64/libpthread.so.0 (0x2aef0d5f2000)
libc.so.6 = /lib64/libc.so.6 (0x2aef0d80e000)
libgssapi_krb5.so.2 = /usr/lib64/libgssapi_krb5.so.2 (0x2aef0db67000)
libkrb5.so.3 = /usr/lib64/libkrb5.so.3 (0x2aef0dd96000)
libcom_err.so.2 = 

Re: [asterisk-users] Problems linking asterisk against self-compiled openssl on CentOS 5

[Alex Villací­s Lasso]

El 23/12/14 a las 12:19, Alex Villací­s Lasso escribió:
I am trying to enable full WebRTC support on asterisk-11.15 for installation on a CentOS 5 machine. Currently the distro cannot be upgraded to any later CentOS series. This CentOS series ships with openssl-0.9.8e, which lacks DTLS-SRTP support required 
for WebRTC. So I decided to build a parallel install of openssl. I chose the Fedora 21 package, openssl-1.0.1j, and built it on CentOS 5. The libraries now reside at /opt/openssl101/usr/lib with header files at /opt/openssl101/usr/include/openssl/ . 
There are symbolic links at /usr/lib64/libssl.so.10 and /usr/lib64/libcrypto.so.10 pointing into my upgraded openssl.


Now I am trying to compile asterisk and link it with my openssl. My configure 
invocation is as follows:

./configure --build=x86_64-redhat-linux-gnu --host=x86_64-redhat-linux-gnu --target=x86_64-redhat-linux-gnu --program-prefix= --prefix=/usr --exec-prefix=/usr --bindir=/usr/bin --sbindir=/usr/sbin --sysconfdir=/etc --datadir=/usr/share 
--includedir=/usr/include --libdir=/usr/lib64 --libexecdir=/usr/libexec --localstatedir=/var --sharedstatedir=/usr/com --mandir=/usr/share/man --infodir=/usr/share/info --with-misdn --with-sounds-cache=no --with-srtp --with-ssl=/opt/openssl101/usr 
--with-crypto=/opt/openssl101/usr


Note the --with-ssl and --with-crypto options at the end, pointing to my 
openssl directory.

After this I compile, but I am getting these messages when compilation reaches 
res/res_rtp_asterisk.c:

a - output/pjlib-x86_64-redhat-linux-gnu/sock_qos_bsd.o
a - output/pjlib-x86_64-redhat-linux-gnu/ssl_sock_common.o
a - output/pjlib-x86_64-redhat-linux-gnu/ssl_sock_ossl.o
a - output/pjlib-x86_64-redhat-linux-gnu/ssl_sock_dump.o
a - output/pjlib-x86_64-redhat-linux-gnu/string.o
a - output/pjlib-x86_64-redhat-linux-gnu/timer.o
a - output/pjlib-x86_64-redhat-linux-gnu/types.o
   [CC] res_rtp_asterisk.c - res_rtp_asterisk.o
res_rtp_asterisk.c: In function 'ast_rtp_dtls_set_configuration':
res_rtp_asterisk.c:1278: warning: implicit declaration of function 
'SSL_CTX_set_tlsext_use_srtp'
res_rtp_asterisk.c: In function 'dtls_srtp_handle_timeout':
res_rtp_asterisk.c:1765: warning: implicit declaration of function 
'DTLSv1_handle_timeout'
res_rtp_asterisk.c: In function 'dtls_srtp_check_pending':
res_rtp_asterisk.c:1817: warning: implicit declaration of function 
'DTLSv1_get_timeout'
res_rtp_asterisk.c: In function 'dtls_srtp_setup':
res_rtp_asterisk.c:1904: warning: implicit declaration of function 
'SSL_export_keying_material'
   [LD] res_rtp_asterisk.o - res_rtp_asterisk.so
   [CC] res_rtp_multicast.c - res_rtp_multicast.o
   [LD] res_rtp_multicast.o - res_rtp_multicast.so

After this, res_rtp_asterisk.so cannot be loaded because it is linked to the 
system openssl but requests the symbols from the upgraded openssl:

[2014-12-22 20:19:41] WARNING[25901] loader.c: Error loading module 
'res_rtp_asterisk.so': /usr/lib64/asterisk/modules/res_rtp_asterisk.so: 
undefined symbol: DTLSv1_handle_timeout

[root@rpmbuild64-2 ~]# ldd /usr/lib64/asterisk/modules/res_rtp_asterisk.so
linux-vdso.so.1 =  (0x7fff431fd000)
libuuid.so.1 = /lib64/libuuid.so.1 (0x2b3f92461000)
libm.so.6 = /lib64/libm.so.6 (0x2b3f92665000)
libnsl.so.1 = /lib64/libnsl.so.1 (0x2b3f928e9000)
librt.so.1 = /lib64/librt.so.1 (0x2b3f92b01000)
libpthread.so.0 = /lib64/libpthread.so.0 (0x2b3f92d0a000)
libcrypto.so.6 = /lib64/libcrypto.so.6 (0x2b3f92f27000)
libssl.so.6 = /lib64/libssl.so.6 (0x2b3f93278000)
libc.so.6 = /lib64/libc.so.6 (0x2b3f934c6000)
/lib64/ld-linux-x86-64.so.2 (0x0037c240)
libdl.so.2 = /lib64/libdl.so.2 (0x2b3f9382)
libz.so.1 = /lib64/libz.so.1 (0x2b3f93a24000)
libgssapi_krb5.so.2 = /usr/lib64/libgssapi_krb5.so.2 (0x2b3f93c38000)
libkrb5.so.3 = /usr/lib64/libkrb5.so.3 (0x2b3f93e67000)
libcom_err.so.2 = /lib64/libcom_err.so.2 (0x2b3f940fc000)
libk5crypto.so.3 = /usr/lib64/libk5crypto.so.3 (0x2b3f942fe000)
libkrb5support.so.0 = /usr/lib64/libkrb5support.so.0 (0x2b3f94524000)
libkeyutils.so.1 = /lib64/libkeyutils.so.1 (0x2b3f9472c000)
libresolv.so.2 = /lib64/libresolv.so.2 (0x2b3f9492e000)
libselinux.so.1 = /lib64/libselinux.so.1 (0x2b3f94b44000)
libsepol.so.1 = /lib64/libsepol.so.1 (0x2b3f94d5c000)
(using libcrypto.so.6 and libssl.so.6 from system openssl)

However, libasteriskssl.so is correctly linked against my upgraded openssl:

[root@rpmbuild64-2 ~]# ldd /usr/lib64/libasteriskssl.so.1
linux-vdso.so.1 =  (0x7fffac39)
libssl.so.10 = /usr/lib64/libssl.so.10 (0x2aef0cfb4000)
libcrypto.so.10 = /usr/lib64/libcrypto.so.10 (0x2aef0d21d000)
libpthread.so.0 = /lib64/libpthread.so.0 (0x2aef0d5f2000)
libc.so.6 = /lib64/libc.so.6 (0x2aef0d80e000)
libgssapi_krb5.so.2 = /usr/lib64/libgssapi_krb5.so.2 (0x2aef0db67000)
libkrb5.so.3 = /usr/lib64

Re: [asterisk-users] asterisk and elastix

[Alex Villací­s Lasso]

El 24/11/14 a las 10:59, Salaheddine Elharit escribió:


Hello list,

i have installed elastix 2.4.0 with call center model and i have created an Outgoing 
Calls https://192.168.1.251/index.php?menu=outgoing_calls my question i want 
to know the name of the tbale where the csv file is uploaded in order to do some 
works.

NB: i found the cdr table in asteriskcdrdb database but the is no information 
related to my csv file



Not exactly the right list. Please repost in the Elastix mailing lists. 
Additionally, please specify yours is a question abount the Elastix Call Center 
module.
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[asterisk-users] OpenSIPS Summit Oct 21st before Astricon

[Alex Goulis]

Hello Everyone!

We wanted to let everyone coming to Astricon know that we will be 
holding an OpenSIPS Summit on Tuesday Oct 21st, 2014 at the Suncoast 
Casino  Spa.


Suncoast is about 10 minutes away from Red Rock and we will be provide 
shuttle service to and from the Summit. For those of you that had to 
book at Suncoast it should be really easy to find us!



Here are some things you can expect...

Ever wonder how you can:

 *

   load balance your Asterisk servers

 *

   provide fail over protection for your Asterisk servers

 *

   distribute services to your Asterisk server clusters

If you have, then you need to attend the OpenSIPS Summit!

This year we'll be including 3 Workshops:

 *

   OpenSIPS, a service enabler for Asterisk

 *

   Scaling Asterisk with OpenSIPS

 *

   OpenSIPS Call Center with Asterisk


Some of the topics this year will include:

 *

   *Flavio Goncalves ***- SIPPulse *- */Using OpenSIPS to protect an
   Asterisk Farm///

 *

   *Pete Kelly - **Sourcevox - *///Solving Business Problems with
   OpenSIPs: Least Cost Routing/

 *

   *Ali Pey ***- Sr. Software Engineer Architect J2 *-
   */o//pensips.cfg///


You can get more information about the summit at:

http://www.opensips.org/Community/Summit-2014LasVegas

Registration is still open:

http://www.opensips.org/Community/Summit-Registration

Last year's summit highlights:

http://www.opensips.org/Community/Summit-2013Atlanta

We're looking forward to seeing you there!


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Re: [asterisk-users] Making sense of SDP for debugging of missing audio in SIP trunk

[Alex Villací­s Lasso]

El 18/06/14 13:44, Alex Villací­s Lasso escribió:
I am debugging an intermittent issue of missing audio on calls that come from a SIP provider into our asterisk-11.10 installation. Sometimes, incoming calls from this provider work correctly, with audio streaming in both directions. Other times, with the 
same setup, the calling party is unable to hear the IVR recording from the asterisk installation, although in fact the streaming is supposed to have started. When debugging with tcpdump, I have seen that all the successful calls have SDP negotiation that 
look like this:


(inside INVITE request body from SIP carrier)
v=0
o=msw.chance4minutes.net 1234 0 IN IP4 38.126.208.46
s=sip call
c=IN IP4 38.126.208.46
t=0 0
m=audio 30552 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20

(inside 200 OK response body from asterisk)
v=0
o=root 835643920 835643920 IN IP4 201.234.196.171
s=Asterisk PBX 11.10.0
c=IN IP4 201.234.196.171
t=0 0
m=audio 12112 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes


In contrast, all failing calls have SDP negotiation that looks like this:

(inside INVITE request body from SIP carrier)
v=0
o=msw.chance4minutes.net 1234 0 IN IP4 38.126.208.46
s=sip call
c=IN IP4 38.126.208.46
t=0 0
m=audio 30534 RTP/AVP 0 18
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=sendrecv

(inside 200 OK response body from asterisk)
v=0
o=root 1719775538 1719775538 IN IP4 201.234.196.171
s=Asterisk PBX 11.10.0
c=IN IP4 201.234.196.171
t=0 0
m=audio 11678 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
a=nortpproxy:yes

Apart from this, I see RTP packets being exchanged in both directions in both 
scenarios.

From what I can see, the SIP provider sends packets that differ in that the 
following four properties are present in the successful case and missing in the 
failing case:
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

What can I deduce from this? Is there some configuration on my asterisk that 
can be tweaked so the failing requests can be handled properly?



Is there additional information needed to make sense of this scenario?



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[asterisk-users] Making sense of SDP for debugging of missing audio in SIP trunk

[Alex Villací­s Lasso]

I am debugging an intermittent issue of missing audio on calls that come from a SIP provider into our asterisk-11.10 installation. Sometimes, incoming calls from this provider work correctly, with audio streaming in both directions. Other times, with the 
same setup, the calling party is unable to hear the IVR recording from the asterisk installation, although in fact the streaming is supposed to have started. When debugging with tcpdump, I have seen that all the successful calls have SDP negotiation that 
look like this:


(inside INVITE request body from SIP carrier)
v=0
o=msw.chance4minutes.net 1234 0 IN IP4 38.126.208.46
s=sip call
c=IN IP4 38.126.208.46
t=0 0
m=audio 30552 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20

(inside 200 OK response body from asterisk)
v=0
o=root 835643920 835643920 IN IP4 201.234.196.171
s=Asterisk PBX 11.10.0
c=IN IP4 201.234.196.171
t=0 0
m=audio 12112 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes


In contrast, all failing calls have SDP negotiation that looks like this:

(inside INVITE request body from SIP carrier)
v=0
o=msw.chance4minutes.net 1234 0 IN IP4 38.126.208.46
s=sip call
c=IN IP4 38.126.208.46
t=0 0
m=audio 30534 RTP/AVP 0 18
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=sendrecv

(inside 200 OK response body from asterisk)
v=0
o=root 1719775538 1719775538 IN IP4 201.234.196.171
s=Asterisk PBX 11.10.0
c=IN IP4 201.234.196.171
t=0 0
m=audio 11678 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
a=nortpproxy:yes

Apart from this, I see RTP packets being exchanged in both directions in both 
scenarios.

From what I can see, the SIP provider sends packets that differ in that the 
following four properties are present in the successful case and missing in the 
failing case:
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

What can I deduce from this? Is there some configuration on my asterisk that 
can be tweaked so the failing requests can be handled properly?

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Re: [asterisk-users] Login by AMI ok, by AJAM fails

[Alex Villací­s Lasso]

El 16/05/14 16:01, Michael L. Young escribió:

- Original Message -

From: Michael L. Young myo...@acsacc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Friday, May 16, 2014 4:55:30 PM
Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails

- Original Message -

From: Michelle Dupuis mdup...@ocg.ca
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Friday, May 16, 2014 4:29:05 PM
Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails

From: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of Michael L.
Young myo...@acsacc.com
Sent: Friday, May 16, 2014 4:16 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails

Have you taken a look at the Wiki yet?

https://wiki.asterisk.org/wiki/display/AST/Allow+Manager+Access+via+HTTP

In looking at that, I see some mistakes in what you are trying to
do.
  Please take a look at that and give it a try.

Well, you need to login first.  Since you are using cURL, you need to
turn the cookie engine on so that it will store and send cookies.

Also, you need to send the login request to
http://localhost:5039/asterisk/manager?action=loginuser=testsecret=test;
and not rawman.  Once you are logged in, then you can get raw
manager output.

I hope that helps.

Michael

Sorry that I messed up the thread while trying to un-top post your message.  
The above was in response to your prior message:


I've done all of that (and I set the AJAM to listen to 5039).  What mistakes do 
you see?

Michael


Sorry if this is already blatantly obvious, but the ampersand character is a 
special character for the shell. The entire URL should be enclosed in single 
quotes in order to send literal ampersands separating the query parameters:

root@apbx:/tmp# curl  
http://localhost:5039/asterisk/rawman?action=loginuser=testsecret=test
[1] 15548
[2] 15549
root@pbx:/tmp# Response: Error
Message: Authentication failed
[1]-  Donecurl 
http://localhost:5039/asterisk/rawman?action=login
[2]+  Doneuser=test

The [1]  and [2]  show that the shell is (incorrectly) spawning 
separate background processes for what it believes are separate commands. Not 
what you want.

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Re: [asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

[Alex Villací­s Lasso]

El 27/04/14 07:47, Barry Flanagan escribió:

On 26 April 2014 00:29, Alex Villací­s Lasso a_villa...@palosanto.com 
mailto:a_villa...@palosanto.com wrote:

I am currently preparing a kamailio-asterisk combination. The asterisk 
installation uses realtime for SIP. The kamailio configuration was based on the 
reference at 
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but 
has
been heavily modified. Currently asterisk runs on localhost and only 
listens on SIP/RTP at 127.0.0.1 . Therefore, all of the SIP traffic appears to 
come from localhost, from the point of view of asterisk.

Currently I have a model on which internal SIP phones get identified by the 
authentication username, and then the contact names at From: and To: get 
massaged to incorporate the SIP domain, in order to emulate multiple-domain 
support. The 'sip' table
in Asterisk defines all such contacts as SIP accounts of the form name_domain.com 
http://name_domain.com, and the SIP phones are configured to use 'name' as 
authentication username for domain 'domain.com http://domain.com'. However, SIP 
providers
that register on the server with authentication names are left with their 
original names, since in the model, SIP trunks are available to all domains.

Now I have to add support for SIP providers which are to be authorized on 
the basis of IP only. Apparently, the kamailio module permissions.so 
(WITH_IPAUTH) is made for just this purpose, so I enabled it. After 
authentication, I need to route the
INVITE to asterisk, and asterisk must somehow match the account for the SIP 
trunk from the available information on the INVITE request.



What I have done in a similar situation is to use  force_send_socket in Kamailio when sending INVITEs from your trusted host (your trunks) so that it is coming in to Asterisk from a different port (say 5070), and then in your Asterisk sip.conf settings 
create a new peer for this like so:


[peer-incoming]
context=peercontext
type=peer
host=127.0.0.1
port=5070

Now, when Asterisk receives an INVITE from 127.0.0.1:5070 http://127.0.0.1:5070 it 
will match this peer, whereas the rest, coming from 127.0.0.1:5060 
http://127.0.0.1:5060, will match your other subscribers.

Here is a bit of the Kamailio config:

if (is_method(INVITE))
{
# If call is coming from a trusted source (Trunk/PSTN) then we send it 
to Asterisk from port 5070
# so that Asterisk knows this is not coming from a subscriber. The peer 
in Asterisk needs to be set with port=5070
# as well as the host=ip address
if (allow_trusted())
{
xlog(L_INFO,Inbound to Asterisk from Trusted Source IP $si, Caller: 
$fU, Callee: $rU with Call-ID $hdr(Call-ID));
force_send_socket(127.0.0.1:5070 http://127.0.0.1:5070);
} else {
# This is a call from a registered subscriber.
xlog(L_INFO,Inbound to Asterisk from $fU to $rU with Call-ID 
$hdr(Call-ID));
}
}
route(RELAY);
exit;
}

NOTE: Kamailio must be set to listen on 127.0.0.1:5070 http://127.0.0.1:5070 as well as your usual ports for this to work! Also, your SIP Trunk trusted peers need to be in the Kamailio trusted table, or explicitly test for the src_ip rather than use 
allow_trusted().



I would rather have a solution that does not involve allocating a new UDP port 
every time a new IP-trusted SIP trunk is configured.

I tried appending a P-Asserted Identity header to the incoming INVITE before 
routing it to asterisk, like this:

#!ifdef WITH_IPAUTH
if((!is_method(REGISTER))  allow_source_address()  $au == )
{
# Attempt to create a P-Asserted-Identity if none exists, to preserve
# incoming Caller-ID
if (!is_present_hf(P-Asserted-Identity))
{
append_hf(P-Asserted-Identity: sip:$fU@$fd\r\n);
}

# Loading $fU from database using IP
sql_pvquery(elxpbx, SELECT name FROM sip WHERE host = '$si' AND sippasswd IS 
NULL, $fU);

# source IP allowed
return;
}
#!endif

With tcpdump, I can see that the header is indeed appended to the SIP headers of the INVITE, but there is no effect in Asterisk. From examination of the Asterisk 11.8.1 source code, I see that channels/chan_sip.c contains a get_pai() function that is 
supposed to process P-Asserted-Identity and extract a caller ID. I am still studying the code, but I would appreciate help on this issue, to see why my attempt is not working.
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Re: [asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

[Alex Villací­s Lasso]

El 02/05/14 10:49, Alex Villací­s Lasso escribió:

El 27/04/14 07:47, Barry Flanagan escribió:

On 26 April 2014 00:29, Alex Villací­s Lasso a_villa...@palosanto.com 
mailto:a_villa...@palosanto.com wrote:

I am currently preparing a kamailio-asterisk combination. The asterisk 
installation uses realtime for SIP. The kamailio configuration was based on the 
reference at 
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but 
has
been heavily modified. Currently asterisk runs on localhost and only 
listens on SIP/RTP at 127.0.0.1 . Therefore, all of the SIP traffic appears to 
come from localhost, from the point of view of asterisk.

Currently I have a model on which internal SIP phones get identified by the 
authentication username, and then the contact names at From: and To: get 
massaged to incorporate the SIP domain, in order to emulate multiple-domain 
support. The 'sip' table
in Asterisk defines all such contacts as SIP accounts of the form name_domain.com 
http://name_domain.com, and the SIP phones are configured to use 'name' as 
authentication username for domain 'domain.com http://domain.com'. However, SIP
providers that register on the server with authentication names are left 
with their original names, since in the model, SIP trunks are available to all 
domains.

Now I have to add support for SIP providers which are to be authorized on 
the basis of IP only. Apparently, the kamailio module permissions.so 
(WITH_IPAUTH) is made for just this purpose, so I enabled it. After 
authentication, I need to route the
INVITE to asterisk, and asterisk must somehow match the account for the SIP 
trunk from the available information on the INVITE request.



What I have done in a similar situation is to use  force_send_socket in Kamailio when sending INVITEs from your trusted host (your trunks) so that it is coming in to Asterisk from a different port (say 5070), and then in your Asterisk sip.conf settings 
create a new peer for this like so:


[peer-incoming]
context=peercontext
type=peer
host=127.0.0.1
port=5070

Now, when Asterisk receives an INVITE from 127.0.0.1:5070 http://127.0.0.1:5070 it 
will match this peer, whereas the rest, coming from 127.0.0.1:5060 
http://127.0.0.1:5060, will match your other subscribers.

Here is a bit of the Kamailio config:

if (is_method(INVITE))
{
# If call is coming from a trusted source (Trunk/PSTN) then we send it 
to Asterisk from port 5070
# so that Asterisk knows this is not coming from a subscriber. The peer 
in Asterisk needs to be set with port=5070
# as well as the host=ip address
if (allow_trusted())
{
xlog(L_INFO,Inbound to Asterisk from Trusted Source IP $si, Caller: 
$fU, Callee: $rU with Call-ID $hdr(Call-ID));
force_send_socket(127.0.0.1:5070 http://127.0.0.1:5070);
} else {
# This is a call from a registered subscriber.
xlog(L_INFO,Inbound to Asterisk from $fU to $rU with Call-ID 
$hdr(Call-ID));
}
}
route(RELAY);
exit;
}

NOTE: Kamailio must be set to listen on 127.0.0.1:5070 http://127.0.0.1:5070 as well as your usual ports for this to work! Also, your SIP Trunk trusted peers need to be in the Kamailio trusted table, or explicitly test for the src_ip rather than use 
allow_trusted().



I would rather have a solution that does not involve allocating a new UDP port 
every time a new IP-trusted SIP trunk is configured.

I tried appending a P-Asserted Identity header to the incoming INVITE before 
routing it to asterisk, like this:

#!ifdef WITH_IPAUTH
if((!is_method(REGISTER))  allow_source_address()  $au == )
{
# Attempt to create a P-Asserted-Identity if none exists, to preserve
# incoming Caller-ID
if (!is_present_hf(P-Asserted-Identity))
{
append_hf(P-Asserted-Identity: sip:$fU@$fd\r\n);
}

# Loading $fU from database using IP
sql_pvquery(elxpbx, SELECT name FROM sip WHERE host = '$si' AND sippasswd IS 
NULL, $fU);

# source IP allowed
return;
}
#!endif

With tcpdump, I can see that the header is indeed appended to the SIP headers of the INVITE, but there is no effect in Asterisk. From examination of the Asterisk 11.8.1 source code, I see that channels/chan_sip.c contains a get_pai() function that is 
supposed to process P-Asserted-Identity and extract a caller ID. I am still studying the code, but I would appreciate help on this issue, to see why my attempt is not working.



By placing debugging statements, I think get_pai() is not being called when receiving an incoming INVITE, corresponding to an incoming call from the IP-authenticated trunk being handled by an IVR, but not yet routed to an internal extension. Why is this 
so? Is this by design?
-- 
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[asterisk-users] SOLVED: Re: Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

[Alex Villací­s Lasso]

El 02/05/14 11:41, Alex Villací­s Lasso escribió:

El 02/05/14 10:49, Alex Villací­s Lasso escribió:

El 27/04/14 07:47, Barry Flanagan escribió:

On 26 April 2014 00:29, Alex Villací­s Lasso a_villa...@palosanto.com 
mailto:a_villa...@palosanto.com wrote:

I am currently preparing a kamailio-asterisk combination. The asterisk 
installation uses realtime for SIP. The kamailio configuration was based on the 
reference at 
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but 
has
been heavily modified. Currently asterisk runs on localhost and only 
listens on SIP/RTP at 127.0.0.1 . Therefore, all of the SIP traffic appears to 
come from localhost, from the point of view of asterisk.

Currently I have a model on which internal SIP phones get identified by the 
authentication username, and then the contact names at From: and To: get 
massaged to incorporate the SIP domain, in order to emulate multiple-domain 
support. The 'sip'
table in Asterisk defines all such contacts as SIP accounts of the form name_domain.com 
http://name_domain.com, and the SIP phones are configured to use 'name' as 
authentication username for domain 'domain.com http://domain.com'. However, SIP
providers that register on the server with authentication names are left 
with their original names, since in the model, SIP trunks are available to all 
domains.

Now I have to add support for SIP providers which are to be authorized on 
the basis of IP only. Apparently, the kamailio module permissions.so 
(WITH_IPAUTH) is made for just this purpose, so I enabled it. After 
authentication, I need to route the
INVITE to asterisk, and asterisk must somehow match the account for the SIP 
trunk from the available information on the INVITE request.



What I have done in a similar situation is to use  force_send_socket in Kamailio when sending INVITEs from your trusted host (your trunks) so that it is coming in to Asterisk from a different port (say 5070), and then in your Asterisk sip.conf settings 
create a new peer for this like so:


[peer-incoming]
context=peercontext
type=peer
host=127.0.0.1
port=5070

Now, when Asterisk receives an INVITE from 127.0.0.1:5070 http://127.0.0.1:5070 it 
will match this peer, whereas the rest, coming from 127.0.0.1:5060 
http://127.0.0.1:5060, will match your other subscribers.

Here is a bit of the Kamailio config:

if (is_method(INVITE))
{
# If call is coming from a trusted source (Trunk/PSTN) then we send it 
to Asterisk from port 5070
# so that Asterisk knows this is not coming from a subscriber. The peer 
in Asterisk needs to be set with port=5070
# as well as the host=ip address
if (allow_trusted())
{
xlog(L_INFO,Inbound to Asterisk from Trusted Source IP $si, Caller: 
$fU, Callee: $rU with Call-ID $hdr(Call-ID));
force_send_socket(127.0.0.1:5070 http://127.0.0.1:5070);
} else {
# This is a call from a registered subscriber.
xlog(L_INFO,Inbound to Asterisk from $fU to $rU with Call-ID 
$hdr(Call-ID));
}
}
route(RELAY);
exit;
}

NOTE: Kamailio must be set to listen on 127.0.0.1:5070 http://127.0.0.1:5070 as well as your usual ports for this to work! Also, your SIP Trunk trusted peers need to be in the Kamailio trusted table, or explicitly test for the src_ip rather than use 
allow_trusted().



I would rather have a solution that does not involve allocating a new UDP port 
every time a new IP-trusted SIP trunk is configured.

I tried appending a P-Asserted Identity header to the incoming INVITE before 
routing it to asterisk, like this:

#!ifdef WITH_IPAUTH
if((!is_method(REGISTER))  allow_source_address()  $au == )
{
# Attempt to create a P-Asserted-Identity if none exists, to preserve
# incoming Caller-ID
if (!is_present_hf(P-Asserted-Identity))
{
append_hf(P-Asserted-Identity: sip:$fU@$fd\r\n);
}

# Loading $fU from database using IP
sql_pvquery(elxpbx, SELECT name FROM sip WHERE host = '$si' AND sippasswd IS 
NULL, $fU);

# source IP allowed
return;
}
#!endif

With tcpdump, I can see that the header is indeed appended to the SIP headers of the INVITE, but there is no effect in Asterisk. From examination of the Asterisk 11.8.1 source code, I see that channels/chan_sip.c contains a get_pai() function that is 
supposed to process P-Asserted-Identity and extract a caller ID. I am still studying the code, but I would appreciate help on this issue, to see why my attempt is not working.



By placing debugging statements, I think get_pai() is not being called when receiving an incoming INVITE, corresponding to an incoming call from the IP-authenticated trunk being handled by an IVR, but not yet routed to an internal extension. Why is this 
so? Is this by design?




SOLVED: asterisk needs to be configured with trustrpid=yes in the affected

[asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

[Alex Villací­s Lasso]

I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has been 
heavily modified. Currently asterisk runs on localhost and only listens on SIP/RTP at 127.0.0.1 . Therefore, all of the SIP traffic appears to come from localhost, from the point of view of asterisk.


Currently I have a model on which internal SIP phones get identified by the authentication username, and then the contact names at From: and To: get massaged to incorporate the SIP domain, in order to emulate multiple-domain support. The 'sip' table in 
Asterisk defines all such contacts as SIP accounts of the form name_domain.com, and the SIP phones are configured to use 'name' as authentication username for domain 'domain.com'. However, SIP providers that register on the server with authentication names 
are left with their original names, since in the model, SIP trunks are available to all domains.


Now I have to add support for SIP providers which are to be authorized on the basis of IP only. Apparently, the kamailio module permissions.so (WITH_IPAUTH) is made for just this purpose, so I enabled it. After authentication, I need to route the INVITE to 
asterisk, and asterisk must somehow match the account for the SIP trunk from the available information on the INVITE request.


A typical INVITE for this scenario looks like this, before being processed by 
kamailio:

INVITE sip:6008010@172.28.161.218:5060;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 
200.25.144.58:5060;branch=z9hG4bK+676ea13f680e853fd847230512a347561+32e3da76+1
Call-ID: FBE75B3A@32e3da76
From: 
sip:042294440@200.25.144.58:5060;user=phone;tag=32e3da76+1+544c000c+52be771c
To: sip:6008010@172.28.161.218:5060;user=phone
CSeq: 975469826 INVITE
Expires: 180
Organization: SetelGYE
Min-SE: 90
Session-Expires: 18000
Supported: replaces, 100rel, timer
Contact: sip:042294440@200.25.144.58:5060;transport=udp;user=phone
Content-Length: 149
Content-Type: application/sdp
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, PRACK, UPDATE, INFO, REFER

v=0
o=- 0 0 IN IP4 201.217.79.3
s=-
c=IN IP4 201.217.79.3
t=0 0
m=audio 5388 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Here, 6008010 is the phone number that was dialed at the provider in order to reach my system, and 042294440 is the provider-supplied Caller-ID, which I want to preserve all the way to Asterisk. In particular, 042294440 appears as the value that ends up as 
$fU (From: username) while being processed in kamailio. If I pass the SIP packet as-is to asterisk, asterisk first tries to match by the value of $fU, which obviously fails to match the trunk name. It then tries to match by incoming IP, which also fails 
because asterisk received this packet from 127.0.0.1 . Finally, asterisk sort of matches to the first record in the sip table, which is *not* the SIP account for this trunk, but some other random account.


I have a partial solution that uses sqlops to make a query to the sip table, using the $si (source IP) and reads the trunk name in order to replace $fU. This works, as now $fU will have the trunk name and asterisk will now recognize the intended SIP 
account for the trunk. However, this has the unfortunate side effect of throwing out the Caller-ID information.


What is the standard/proper way to deal with this situation? Is there a well-known way to make Asterisk match the trunk name, without overwriting the Caller-ID information? Before you ask, requesting the provider to modify its INVITEs is not an option. I 
believe there is a standard way to deal with this, since this scenario should also arise with a kamailio that faces the internet, and relays INVITEs (after authentication) to an asterisk in a local network. As far as I can tell, the fact that in my case 
the 'local network' is localhost should be irrelevant.
-- 
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   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

[Alex Villací­s Lasso]

El 25/04/14 18:29, Alex Villací­s Lasso escribió:
I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has been 
heavily modified. Currently asterisk runs on localhost and only listens on SIP/RTP at 127.0.0.1 . Therefore, all of the SIP traffic appears to come from localhost, from the point of view of asterisk.


Currently I have a model on which internal SIP phones get identified by the authentication username, and then the contact names at From: and To: get massaged to incorporate the SIP domain, in order to emulate multiple-domain support. The 'sip' table in 
Asterisk defines all such contacts as SIP accounts of the form name_domain.com, and the SIP phones are configured to use 'name' as authentication username for domain 'domain.com'. However, SIP providers that register on the server with authentication 
names are left with their original names, since in the model, SIP trunks are available to all domains.


Now I have to add support for SIP providers which are to be authorized on the basis of IP only. Apparently, the kamailio module permissions.so (WITH_IPAUTH) is made for just this purpose, so I enabled it. After authentication, I need to route the INVITE 
to asterisk, and asterisk must somehow match the account for the SIP trunk from the available information on the INVITE request.


A typical INVITE for this scenario looks like this, before being processed by 
kamailio:

INVITE sip:6008010@172.28.161.218:5060;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 
200.25.144.58:5060;branch=z9hG4bK+676ea13f680e853fd847230512a347561+32e3da76+1
Call-ID: FBE75B3A@32e3da76
From: 
sip:042294440@200.25.144.58:5060;user=phone;tag=32e3da76+1+544c000c+52be771c
To: sip:6008010@172.28.161.218:5060;user=phone
CSeq: 975469826 INVITE
Expires: 180
Organization: SetelGYE
Min-SE: 90
Session-Expires: 18000
Supported: replaces, 100rel, timer
Contact: sip:042294440@200.25.144.58:5060;transport=udp;user=phone
Content-Length: 149
Content-Type: application/sdp
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, PRACK, UPDATE, INFO, REFER

v=0
o=- 0 0 IN IP4 201.217.79.3
s=-
c=IN IP4 201.217.79.3
t=0 0
m=audio 5388 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Here, 6008010 is the phone number that was dialed at the provider in order to reach my system, and 042294440 is the provider-supplied Caller-ID, which I want to preserve all the way to Asterisk. In particular, 042294440 appears as the value that ends up 
as $fU (From: username) while being processed in kamailio. If I pass the SIP packet as-is to asterisk, asterisk first tries to match by the value of $fU, which obviously fails to match the trunk name. It then tries to match by incoming IP, which also 
fails because asterisk received this packet from 127.0.0.1 . Finally, asterisk sort of matches to the first record in the sip table, which is *not* the SIP account for this trunk, but some other random account.


I have a partial solution that uses sqlops to make a query to the sip table, using the $si (source IP) and reads the trunk name in order to replace $fU. This works, as now $fU will have the trunk name and asterisk will now recognize the intended SIP 
account for the trunk. However, this has the unfortunate side effect of throwing out the Caller-ID information.


What is the standard/proper way to deal with this situation? Is there a well-known way to make Asterisk match the trunk name, without overwriting the Caller-ID information? Before you ask, requesting the provider to modify its INVITEs is not an option. I 
believe there is a standard way to deal with this, since this scenario should also arise with a kamailio that faces the internet, and relays INVITEs (after authentication) to an asterisk in a local network. As far as I can tell, the fact that in my case 
the 'local network' is localhost should be irrelevant.




If I manage to coax Kamailio to add a (synthetized) P-Asserted-Identity header 
to the INVITE request before sending it to Asterisk, will Asterisk be able to 
use it? Will this information show up on a CDR?
-- 
_
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   http://www.asterisk.org/hello

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To UNSUBSCRIBE or update options visit:
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[asterisk-users] Asterisk SSL support broken with update from openssl-1.0.0 to 1.0.1e, recompiling does *not* help

[Alex Villací­s Lasso]

I am having an issue that prevents WebSockets over SSL/TLS (or any kind of 
encrypted HTTP traffic to Asterisk) from working after an openssl library 
update.

My setup is CentOS 6 x86_64, and initially, with openssl[-devel]-1.0.0-20.el6_2.5.x86_64 . With this openssl versions, https over TCP port 8089 initializes correctly with asterisk-11.7.0. After an upgrade to openssl[-devel]-1.0.1e-16.el6_5.4.x86_64 , I 
compiled asterisk-11.8.1 . When testing the exact same configuration, I noticed that TCP port 8089 was no longer listening, even though TCP port 8088 (standard unencrypted HTTP) was. After a patch coaxing some error message to be shown in the logs, I got 
the following:


[Mar 27 10:25:47] DEBUG[10516] config.c: Parsing /etc/asterisk/acl.conf
[Mar 27 10:25:47] VERBOSE[10516] config.c:   == Parsing 
'/etc/asterisk/acl.conf': Found
[Mar 27 10:25:47] DEBUG[10516] config.c: Parsing /etc/asterisk/http.conf
[Mar 27 10:25:47] VERBOSE[10516] config.c:   == Parsing 
'/etc/asterisk/http.conf': Found
[Mar 27 10:25:47] DEBUG[10516] netsock2.c: Splitting '0.0.0.0' into...
[Mar 27 10:25:47] DEBUG[10516] netsock2.c: ...host '0.0.0.0' and port ''.
[Mar 27 10:25:47] DEBUG[10516] config.c: extract uint from [8088] in [0, 65535] 
gives [8088](0)
[Mar 27 10:25:47] DEBUG[10516] netsock2.c: Splitting '0.0.0.0:8089' into...
[Mar 27 10:25:47] DEBUG[10516] netsock2.c: ...host '0.0.0.0' and port '8089'.
[Mar 27 10:25:47] DEBUG[10516] config.c: extract addr from 0.0.0.0:8089 gives 
0.0.0.0:8089(0)
[Mar 27 10:25:47] VERBOSE[10516] http.c:  Bound HTTP server to address 0.0.0.0:0
[Mar 27 10:25:47] DEBUG[10516] tcptls.c: Sorry, SSL_CTX_new call returned null... 
(sslerror=336236705 sslstring=error:140A90A1:lib(20):func(169):reason(161))
[Mar 27 10:25:47] VERBOSE[10516] manager.c:   == Manager registered action Ping

From googling around, I see that reason(161) means that somehow, there are no loaded cyphers for SSL (source: http://marc.info/?l=openssl-usersm=99129794725716w=2). This source also claims that the most common cause is that SSL_library_init() was not 
called. However, I do not see anything in the Asterisk changelog that could possibly cause the call to be skipped.


I see nothing out of the ordinary when configuring Asterisk using the updated 
openssl:

checking sqlite3.h usability... yes
checking sqlite3.h presence... yes
checking for sqlite3.h... yes
checking for AES_encrypt in -lcrypto... yes
checking openssl/aes.h usability... yes
checking openssl/aes.h presence... yes
checking for openssl/aes.h... yes
checking for SSL_connect in -lssl... yes
checking openssl/ssl.h usability... yes
checking openssl/ssl.h presence... yes
checking for openssl/ssl.h... yes
checking osp/osp.h usability... no
checking osp/osp.h presence... no
checking for osp/osp.h... no
checking for SSL_CTX_set_tlsext_use_srtp in -lssl... yes
checking for openssl/ssl.h... (cached) yes
checking for srtp_init in -lsrtp... yes
checking srtp/srtp.h usability... yes
checking srtp/srtp.h presence... yes
checking for srtp/srtp.h... yes
checking for the ability of -lsrtp to be linked in a shared object... yes
checking for srtp_shutdown in -lsrtp... yes
checking for srtp/srtp.h... (cached) yes
checking for malloc in -lhoard... no
checking for dbinit in -lsybdb... yes
checking sybdb.h usability... yes
checking sybdb.h presence... yes

What else could I check? I have checked issues.asterisk.org and there is no 
mention of a situation similar to mine, so I believe this is somehow related to 
the build environment.

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[asterisk-users] SOLVED: Re: Asterisk SSL support broken with update from openssl-1.0.0 to 1.0.1e, recompiling does *not* help

[Alex Villací­s Lasso]

El 27/03/14 11:59, Alex Villací­s Lasso escribió:

I am having an issue that prevents WebSockets over SSL/TLS (or any kind of 
encrypted HTTP traffic to Asterisk) from working after an openssl library 
update.

My setup is CentOS 6 x86_64, and initially, with openssl[-devel]-1.0.0-20.el6_2.5.x86_64 . With this openssl versions, https over TCP port 8089 initializes correctly with asterisk-11.7.0. After an upgrade to openssl[-devel]-1.0.1e-16.el6_5.4.x86_64 , I 
compiled asterisk-11.8.1 . When testing the exact same configuration, I noticed that TCP port 8089 was no longer listening, even though TCP port 8088 (standard unencrypted HTTP) was. After a patch coaxing some error message to be shown in the logs, I got 
the following:
Sorry about the noise. Turns out, a coworker was testing an experimental patch to asterisk and compiled on a machine without openssl-devel installed, then left a stray dummy /usr/lib/libasteriskssl.so.1 that was overriding the one at /usr/lib64 until just 
a minute ago.


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Re: [asterisk-users] How to configure asterisk to only accept SIP from kamailio@localhost but exchange RTP on all interfaces?

[Alex Villací­s Lasso]

El 25/02/14 08:30, Karsten Wemheuer escribió:

Hi Alex,

Am Donnerstag, den 20.02.2014, 13:48 -0500 schrieb Alex Villací­s Lasso:

I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following
the setup guide at
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I 
want to run asterisk and kamailio on the same server, with SIP realtime 
configuration
(MySQL database) so that kamailio authenticates and then forwards the
registration to asterisk on localhost. The setup calls for asterisk to
be configured to listen for SIP traffic on all interfaces, on a
nonstandard port (I chose 5080). It also calls for
blanking of the password for the SIP peer (in my case, a softphone),
so that it will not request for authentication again. I have managed
to make a call with working audio from the softphone to an extension
on asterisk through kamailio.

My concern is that asterisk is left listening for SIP through all
interfaces and with no SIP passwords. I want to secure the setup
against directed traffic to the asterisk UDP port (5080), that
bypasses the kamailio process. I tried setting
bindaddr=127.0.0.1 so asterisk will only listen for SIP traffic on
localhost, but this has the side effect of also removing audio - the
call appears to be successful on the softphone and on the asterisk
logs, but no audio is actually heard. My theory is
that the RTP traffic is being sent to kamailio instead of the
softphone.

How can I set up asterisk so that it can send RTP anywhere but reject
any SIP traffic that does not come from the kamailio process on
localhost?


If You bind asterisk to 127.0.0.1 I think the media connection is set
for this IP. Your Softphone can not reach the correct 127.0.0.1
(localhost is everywhere).

I would suggest, You setup asterisk on eth0 address or 0.0.0.0. In the
sip.conf You could secure Your setup with
 deny = 0.0.0.0/0.0.0.0
 permit = Your-LAN-Adress
This way asterisk accepts SIP from Your box only.


This might work, but would need to touch sip.conf every time the IP address 
changes. It would be nice to have a configuration that can be set up once and 
not modified again. That is why I wanted to set up localhost.

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[asterisk-users] How to configure asterisk to only accept SIP from kamailio@localhost but exchange RTP on all interfaces?

[Alex Villací­s Lasso]

I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configuration 
(MySQL database) so that kamailio authenticates and then forwards the registration to asterisk on localhost. The setup calls for asterisk to be configured to listen for SIP traffic on all interfaces, on a nonstandard port (I chose 5080). It also calls for 
blanking of the password for the SIP peer (in my case, a softphone), so that it will not request for authentication again. I have managed to make a call with working audio from the softphone to an extension on asterisk through kamailio.


My concern is that asterisk is left listening for SIP through all interfaces and with no SIP passwords. I want to secure the setup against directed traffic to the asterisk UDP port (5080), that bypasses the kamailio process. I tried setting 
bindaddr=127.0.0.1 so asterisk will only listen for SIP traffic on localhost, but this has the side effect of also removing audio - the call appears to be successful on the softphone and on the asterisk logs, but no audio is actually heard. My theory is 
that the RTP traffic is being sent to kamailio instead of the softphone.


How can I set up asterisk so that it can send RTP anywhere but reject any SIP 
traffic that does not come from the kamailio process on localhost?

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Re: [asterisk-users] How to configure asterisk to only accept SIP from kamailio@localhost but exchange RTP on all interfaces?

[Alex Villací­s Lasso]

El 20/02/14 15:07, Markus escribió:

Am 20.02.2014 19:48, schrieb Alex Villací­s Lasso:

My concern is that asterisk is left listening for SIP through all
interfaces and with no SIP passwords. I want to secure the setup against
directed traffic to the asterisk UDP port (5080), that bypasses the
kamailio process. I tried setting bindaddr=127.0.0.1 so asterisk will
only listen for SIP traffic on localhost, but this has the side effect
of also removing audio - the call appears to be successful on the
softphone and on the asterisk logs, but no audio is actually heard. My
theory is that the RTP traffic is being sent to kamailio instead of the
softphone.


Theories are nice, but you should check whether they are true using, e.g., 
tcpdump :)

I would check with, for example:

tcpdump -nnnqt -s 0 -A -i eth0 port 5060

or instead of port 5060 (or 5080) try udp to see what is going on with RTP. Change from eth0 to lo to see if there is really RTP going to nowhere. When looking at port 5060/5080, check the SDP header to see what kamilio/Asterisk/your softphone 
announce in terms of RTP.


I thought kamailio is a SIP server/proxy only and is not involved in RTP at all.



From a wireshark capture, what the softphone sees when contacting kamailio (in 
the bindaddr=127.0.0.1 configuration) is that the media negotiation in the OK 
package contains a random UDP port (as expected) but indicates that the IP for 
RTP is 127.0.0.1 .

I have also tried deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.0.0.0 , but the 
softphone then gets denied through the kamailio too. It seems deny/permit 
restricts the IP of the original contact, not the IP the SIP traffic was 
received through.


In any case, if you want to only allow only certain connections from somewhere 
to somewhere (including from/to certain ports), iptables is your friend if you 
are using Linux.


I know iptables would solve my issue, and I will certainly use it, but I do not want to rely on iptables as the *only* thing that prevents the sip proxy bypass. I want an asterisk configuration that will only accept SIP signaling traffic coming from a 
specific IP (in my case, 127.0.0.1), but will then negotiate RTP across any interface that will contact the softphone. I want this to work with the IP of the original contact.


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Re: [asterisk-users] Asterisk 11.7.0 not receiving registration from local address

[Alex Villací­s Lasso]

El 22/01/14 08:39, Administrator TOOTAI escribió:

Le 22/01/2014 14:01, Administrator TOOTAI a écrit :

Hi,

I face a problem which look like the same as David with his thread Asterisk not 
receiving call from VPN address.

I had an Asterisk (Elastix) working well in a VM (Debian Wheezy - KVM) having 
IP 192.168.111.14, my phone network is in the range 192.168.10.x

I updated lately to 11.7.0 version and now no one of my phones can register anymore 
to the asterisk. Ngrep as well as wireshark shows the traffic going in on eth0 from 
VM, but inside Asterisk, nothing, sip set debug ip ip from phones shows 
nothing.

This asterisk is also connected to other trunks we have outside our network, 
there is no problem here, registration is fine. Problem seems to be only with 
intranet addresses.

We have localnet=192.168.0.0/255.255.0.0, We add permit=0.0.0.0/0.0.0.0, 
nothing to do :-(

Thanks for any hint



Definitely something wrong with 11.7.0 I switched back to 11.6.0.1 and phone 
registration is OK.

David, you should probably try to downgrade if you have an 11.7.0 installed


In our local tests, we have seen no special issues regarding SIP registration 
when updating to 11.7.0.

Can you reproduce the issue on a separate virtual machine with just one 
extension and a separate install of Elastix updated to asterisk-11.7.0 ?
If at all possible, can you reproduce the issue on a physical machine with 
Elastix installed?

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[asterisk-users] Transfer call placed from console (with chan_alsa)

[Alex]

Hi everyone.

Having experimented a but with a prototype of a system I described in
an earlier thread (Reading DTMF sent by callee during a SIP call), I
decided to implement my requirement by transferring the call to
another extension. This way, the callee can open the door by pressing
#1, and the dial plan for extension 1 takes care of the rest.

This works when I make a typical SIP to SIP call, but it doesn't when
I call from the console, using chan_alsa. I can see that the transfer
feature is inactive:

rasterisk*CLI core show channeltype console
-- Info about channel driver: Console --
  Device State: no
Indication: yes
 Transfer : no
  Capabilities: 0x40 (slin)
   Digit Begin: no
 Digit End: yes
Send HTML : no
 Image Support: no
  Text Support: yes



However, I am unable to find a way to activate it. How can I transfer
placed from the console? Is it possible, in principle?


Alex

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Re: [asterisk-users] Reading DTMF sent by callee during a SIP

[Alex]

Don,
 Isn't it easier just to use a SIP door phone?
Not in my case. The purpose of the whole exercise is to turn it into a
DIY project and hopefully, train a bunch of kids how to tinker with
RaspberryPi, Linux and Asterisk. Therefore, my objective is not to
give a fish, but to teach how to fish and make them able to see
the fun in it :-)


Ish,
 You could create your own feature in features.conf that executes a
 Macro/Gosub defined in sip.conf...
Hmmm, I have never dealt with customized features.conf in the past, so
I will look into that direction.



So far, my approach is to leverage what I already know - and that is
to simply transfer the call to a specific extension; which, in turn -
will call System() in order to send a command to the electro-magnet
that keeps the door locked.

The downside is that one would have to press #5 (assuming that 5 is
that magic extension for opening the door), rather than just #.



Alex

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[asterisk-users] Reading DTMF sent by callee during a SIP call

[Alex]

Hi everyone,

I am looking for advice about the design of a SIP-based intercom. I
count on your help, as my current attempts are not fruitful (yet).

This will be a pretty long message, so here's my fundamental question:

Is there a way to interpret DTMF tones sent by the calee
(not the caller) while a voice call is in progress?






Here's the desired scenario:

- there is a box with speakers and a mic
- Asterisk is running on a computer inside that box
- the box is embedded in a door
- There are two user accounts, UserA and userB
- UserA is a client that runs on the server*
- UserA calls UserB and they are having a voice conversation


Throughout the call, Asterisk must react to DTMF tones sent by userB;
such that an action is executed when a specific key is pressed.

The idea is to build an intercom that would enable me to open a door
remotely, by relying entirely on SIP, so there would be no need to
have some additional communication channel to send the open door
signal.




I have previously implemented IVRs using `Background` and jumped to
specific extensions, when a button was pressed. But in that case, the
extensions are dialed by the caller; whereas now the input must from
the person who answered the call.

If I use `Dial` and `Read` - the latter is only executed after `Dial`
terminates - so this is not suitable.


`Background` behaves like I need - but it plays back a predefined
file, so it is not suitable for an interactive conversation.



* Having a SIP client on the same machine as the Asterisk server
itself is not possible, because both won't be able to bind to port
5060. My guess is that the solution is to originate a call from the
CLI; but I haven't gotten to that part yet.




Thank you for your patience, I am looking forward to your feedback,
Alex

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[asterisk-users] How do I remotely force an *unconfigured* Digium DPMA phone to re-query the network for the DPMA server?

[Alex Villací­s Lasso]

Consider the following scenario:

1) One or more Digium DPMA phones are plugged into the network. I know their IP 
addresses and MACs.
2) The Asterisk I want to use as the telephony server starts without the DPMA 
module. Therefore there are no DPMA sessions between the phones and the server.
3) I install DPMA on the server, and write its configuration file for the 
phones. I will tie each phone to its configuration using the MAC, which I 
already know (config_auth=mac)
4) I reload the module as required to reload the configuration file.

Now that everything else is ready, I have to make the previously unconfigured 
Digium phones to query the net for the DPMA server.

How?

I would rather not physically locate and reset every single phone (all 100 of them, to cite a random number). I want a way to remotely force the phones to re-query the net and discover the configured DPMA server. The digium_phones reconfigure all would 
work for this, except that it needs an existing DPMA session, and there is none, since I configured the server for the first time. However, since I know the IP addresses of the phones, I could send some kind of notification to the phones. But I do not know 
which one. For some phones other than Digium, some variations of the SIP NOTIFY command will force the phone to request its configuration anew. Is there any equivalent for Digium phones?


A variation: the same Asterisk server has a bunch of configured DPMA phones. I now add a fresh batch of phones to the net. Since their MACs are unknown to the Asterisk server, the phones will remain unconfigured. I add the MACs to the file and reload the 
module. How do I make the fresh batch of phones to re-query the net and pull its configuration without having to physically go to each one and reset or otherwise interact with it?


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Re: [asterisk-users] DPMA, check-sync

[Alex Villací­s Lasso]

El 06/09/13 14:44, Malcolm Davenport escribió:

Howdy,

Please forgive the off-list e-mail.  I'm not subscribed to the list, I only 
peruse the archives.

The follow up from George is correct.  For phones that have already been 
attached to DPMA, DPMA disables the enable_check_sync phone setting.

For phones that have not yet been attached to DPMA, that's not the case, as 
enable_check_sync is enabled out of the box.  So, a properly formatted SIP 
NOTIFY will do the trick.

You can do that outside of Asterisk using sipp or sipsak or something similar.

To cover the case for phones that are already attached to DPMA, we'll add a new phone option in DPMA 1.7.1.  We're on DPMA 1.6.0 right now, and DPMA 1.7.0 is already in the latter stages of internal QA.  I don't want to derail that train as we're very 
nearly there.


The new phone option will simply expose the ability to turn enable_check_sync 
on or off for phones.  We'll still default to the current behavior of disabling 
it though for phones attached to DPMA.

Apologies for the inconvenience.


The check-sync notify does not work in all required scenarios. I tracked the 
traffic with wireshark. To demonstrate:

Start from a D40 phone, factory resetted. The asterisk server starts with DPMA 
loaded but with a zero-bytes conf.
Write a valid configuration that includes the phone.
On Asterisk: module reload res_digium_phone.so ; digium_phones reconfigure all 
; sip notify CHECKSYNC $PHONE_IP
(this makes the phone re-query the network in this one time)
(phone gets reconfigured and registers the SIP accounts)
Now, remove the phone section, but keep all other sections intact.
On Asterisk: module reload res_digium_phone.so ; digium_phones reconfigure all 
; sip notify CHECKSYNC $PHONE_IP
(phone had a DPMA session, so it reloads configuration)
(phone tries to fetch configuration, but fails, as expected. The phone gets stuck in the 
Communication Error screen)
Now, restore the phone section so the configuration returns to the previously 
working state.
On Asterisk: module reload res_digium_phone.so ; digium_phones reconfigure all 
; sip notify CHECKSYNC $PHONE_IP
(sip notify gets sent, phone answers with 200 Ok)
(nothing happens. The phone is still stuck at the Communication Error screen)

What should happen is that the phone should re-scan at the last step and reload 
the configuration, as it did the first time.

The CHECKSYNC mentioned above is any entry in sip_notify.conf that references check-sync. 
As mentioned in the mailing list, I already know the phone IP for the sip 
notify command.

What I think is happening is that if the phone sees a DPMA server but the server has no valid configuration for the phone, the phone gets stuck in a state where check-resync will not make the phone rescan an updated configuration, and needs to be reset 
manually.


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Re: [asterisk-users] LibopenR2 with debug symbols

[Alex Villací­s Lasso]

El 05/09/13 07:02, Rodrigo Montiel escribió:

Hi all,

Recently I have raised up a bug (related to segmentation fault of asterisk 
process) due to a core dump generated by Asterisk under the following 
environment:

CentOS release 5.7 (Final)
Kernel 2.6.18-238.12.1.el5 #1 SMP Tue May 31 13:23:01 EDT 2011 i686 i686 i386 
GNU/Linux
Asterisk 1.8.23.0
DAHDI Version: 2.6.1 Echo Canceller: HWEC, OSLEC
libpri-1.4.14-0
libpri-devel-1.4.14-0
libopenr2-1.3.2-1
libopenr2-devel-1.3.2-1
Cards Installed: 03:08.0 Communication controller: Digium, Inc. Wildcard TE420 
quad-span T1/E1/J1 card 3.3V (PCI-Express) (5th gen) (rev 02)

A piece of gdb output looks like this:

Core was generated by `/usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c'.
Program terminated with signal 11, Segmentation fault.
#0  0x0014c4e5 in vfprintf () from /lib/libc.so.6
#0  0x0014c4e5 in vfprintf () from /lib/libc.so.6
No symbol table info available.
#1  0x00156742 in fprintf () from /lib/libc.so.6
No symbol table info available.
#2  0x00949d59 in ?? () from /usr/lib/libopenr2.so.3
No symbol table info available.
#3  0x in ?? ()
No symbol table info available.

The thing is that Asterisk development guys are poninting to a problem on 
libopenR2 library, and asked me to install library with debug symbols.
I am newby on this, can anyone there help me on how to mix debug symbols on 
installation of openR2? Did anyone experience this issue?

I tried googling but no success, sorry if my question is very basic :(


Is this an Elastix system? If so, then run:

yum update libopenr2*
yum install libopenr2-debuginfo

This should update libopenr2 to 1.3.3 *and* install the debuginfo package. Then 
you should try to reproduce the crash.
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[asterisk-users] Use DPMA to enumerate unconfigured Digium phones in LAN

[Alex Villací­s Lasso]

Is there a way to use DPMA to enumerate the Digium phones that are plugged in and visible in the local network, but not (yet) configured through the DPMA configuration files in Asterisk. I would like to write a frontend that lists the DPMA capable phones, 
presents a GUI to specify the various options, then write the configuration files as required and make the phones read these settings. Ideally such a report would indicate the MAC address for the phone and preferably the current IP.


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[asterisk-users] Need to figure out DAHDI logical group from CDR record

[Alex Villací­s Lasso]

I have a bunch of CDR records in the mysql database asteriskcdrdb on a FreePBX system. There is a DAHDI trunk defined in FreePBX which uses the gN identifier to make calls. So in this setup the trunk is roughly equivalent to a DAHDI logical group. I 
want to know, given a CDR, which logical group (and therefore, trunk) was used to place the call, or was used to receive the call.


In Asterisk 1.6 and lower, the DAHDI channel encoded the B channel, as in DAHDI/aaa-bbb where aaa was the ISDN B channel. Since, as far as I can see, the logical group groups channels together, the strategy was to ask Asterisk for the dahdi groups, list 
the B channels, and fetch the records that referenced those B channels.


In Asterisk 1.8, this no longer works. Since the DAHDI channel name no longer references a B channel ( https://issues.asterisk.org/jira/browse/ASTERISK-17229) I can no longer extract the B channel to associate a particular CDR with a particular logical 
group, Though, I am not particularly interested in the B channel, except as a means to build the association.


Is there a way to guess the logical group given the standard CDR fields? All of 
the strategies I can think of require adding new fields to asteriskcdrdb, which 
does nothing to help me with the CDR records already written.

The strategies for future association involve using the dialplan to capture CHANNEL(dahdi_span/dahdi_channel/dahdi_type) and writing this information in additional fields in asteriskcdrdb. However, some field to inform the logical group would be more 
useful for me.
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[asterisk-users] Asterisk 11, SIP. OK to BYE goes to wrong ip/port combination

[Alex Zarubin]

Hi all,

I've read several discussions about asterisk adding 'received' parameter to the 
top Via header.

In our case asterisk (release 11.4) gets BYE from sip proxy (with BYE top via 
header containing proxy ip address and port) but added 'received' parameter 
contains ip address from a 2nd Via (or from From') and OK gets lost.

I'm just trying to adjust sip configuration that used to work for simple call 
scenarios (in 1.4, for example) for Asterisk 11.
Your input is appreciated.

Thank you.

Alex Zarubin


In sip.conf

[general]
nat = no
outboundproxy=PROXYipaddress:PROXYport

[CARRIER]
type=peer
host=CARRIERipaddress
port=CARRIERport
canreinvite=no

Outbound call from asterisk is established normally via outbound proxy. BYE 
coming from the CARRIER

--- SIP read from UDP:PROXYipaddress:PROXYport ---
BYE sip:XYZ@ASTERISKipaddress:5060 SIP/2.0
Via: SIP/2.0/UDP PROXYipaddress:PROXYport;branch=z9hG4bK-whatever-rsMNRDPbOY.0
Via: SIP/2.0/UDP 
CARRIERipaddress:CARRIERport;branch=z9hG4bK+17d247e1b781cd07f8d5339588fb32091+127.0.0.1+1
From: 
sip:ABC@CARRIERipaddress:CARRIERport;tag=127.0.0.1alUtKGp-07233+1+9362fc7+68f1babf
To: sip:BLAH@OURipaddress;tag=as1ca814af
Call-ID: 146b538429a2380a68bb374543d40c6d@OURipaddress:5060
CSeq: 1035426164 BYE
Max-Forwards: 69
User-Agent: Alcatel-Lucent 5060 MGC-8 8.3.0.6.SP1.2
Content-Length: 0
Supported: replaces, 100rel

Asterisk adds received=CARRIERipaddress (taken either from 2nd Via or from 
'From') and sends OK to CARRIERipaddress:PROXYport.
This OK goes nowhere, carrier re-sends BYE several times...

--- Transmitting (no NAT) to CARRIERipaddress:PROXYport ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
PROXYipaddress:PROXYport;branch=z9hG4bK-whatever-rsMNRDPbOY.0;received= 
CARRIERipaddress
Via: SIP/2.0/UDP 
CARRIERipaddress:CARRIERport;branch=z9hG4bK+17d247e1b781cd07f8d5339588fb32091+127.0.0.1+1
From: 
sip:ABC@CARRIERipaddress:CARRIERport;tag=127.0.0.1alUtKGp-07233+1+9362fc7+68f1babf
To: sip:BLAH@OURipaddress;tag=as1ca814af
Call-ID: 146b538429a2380a68bb374543d40c6d@OURipaddress:5060
CSeq: 1035426164 BYE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Content-Length: 0


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Re: [asterisk-users] how to launch a URl when dialing a number

[Alex Villací­s Lasso]

El 31/05/13 09:21, Salaheddine Elharit escribió:

thanks justin i try to do this but the issue still the same.this link is stored 
in my server 192.168.5.109 .but what i want to receive this link when i call 
this number in my pc

ip adresse of my pc 192.168.5.131
ip adresse of server when the page php is stored

thanks and regards



2013/5/30 Justin Killen jkil...@allamericanasphalt.com 
mailto:jkil...@allamericanasphalt.com

If you just want the url to be opened (perhaps to update a counter via a 
web service or cgi script), you can do this:

system(wget http://;)

or

system(fetch http://...;)

-Justin




*From:*asterisk-users-boun...@lists.digium.com 
mailto:asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com 
mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Salaheddine Elharit
*Sent:* Thursday, May 30, 2013 8:07 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] how to launch a URl when dialing a number

Hello

i want to luanch an URL in my PC when i call a number  like below

exten = 066104,1,Set(CALLERID(number)=52xxx)

exten = 066104,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))

exten = 
066104,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))

exten = 066104,n,http://192.168.5.109/interface2/interface2.php ( here 
i want to launch this url in my pc )

exten = 066104,n,Hangup()




From this discussion, I am guessing the following scenario. Please correct me 
if I am wrong.
- There are (at least) three roles in your scenario: the Asterisk server, the 
PHP webserver (which may or may not be the same machine as the Asterisk 
server), and the client PC.
- Apparently your client PC runs a softphone (but the exact nature of the 
telephony client is not important).
- A call is connected from the phone to your Asterisk, is directed to your 
context, and dials some trunk (Zap/g1 in your snippet).
- You then want, somehow, to make the Asterisk server reach out to your client 
PC (which runs a GUI and has a web browser) and force it to open an arbitrary 
web page on the PHP webserver, presumably a callcenter data collecting form.

The problematic issue is the last part. Especially the implication of remotely 
opening a web page on some random PC.

If the above scenario is in fact what you were planning to do, maybe you need to rethink your design. In the default case, there is no way to make a remote PC open an arbitrary URL on its GUI. Think about the security implications. You should instead have 
the web interface already open, and program a Click2Call capability that contacts the Asterisk server and uses AMI to execute an Originate action with your context as your target. Then the web page would load your target URL in order to handle the call. 
Or, if the calls come from an external source, you should program some kind of monitor that alerts the web interface that the call was handled by the context.
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Re: [asterisk-users] Not able to build the chan_sip.c module

[Alex Villací­s Lasso]

El 27/05/13 01:56, upendra escribió:

Hi,

i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i see  in make menuselect options it showing XXX -- extended , please let me know how to enable it and 
make build chan_sip module.



Why are you trying to build asterisk manually on the Elastix machine? Have you 
discussed your needs in the Elastix forums? If so required, you can try to 
tweak the asterisk source RPM from the Elastix project.

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Re: [asterisk-users] How to allow AMI access to Originate yet deny Application: System

[Alex Villací­s Lasso]

El 15/05/13 10:10, Alex Villací­s Lasso escribió:
While doing a security audit on a system I maintain, I stumbled upon an unvalidated use of a variable to compose an Originate request to the local Asterisk instance via AMI. Taking as an example an earlier exploit for FreePBX, I realized that this, 
combined with Application: System as an injected value, could allow arbitrary code execution. I am in the process of fixing all instances of this bug in our system. However, there are third parties that plug into our system, and that reconfigure the 
manager.conf file to allow remote access to AMI logins that allow Originate (by default, the manager.conf remains configured to deny login to any system except localhost). I want to have a guideline on how to proceed in order to make these applications 
work, without allowing malicious users to compromise the system. I know that one way to proceed is to deny remote access to AMI, and build an application-specific proxy that will perform the Originate on behalf of the remote requester, after filtering 
the values. However, I want to know if there is a simpler way to remove the danger of code execution while allowing applications to use AMI to place calls.


The intended scenario is that a remote desktop application (for Windows) is configured with the AMI credentials, and connects over the LAN to Asterisk in order to place calls and otherwise monitor the system. The attack I want to protect against is that 
of a malicious user that collects the credentials from the desktop application and proceeds to use the Application: System trick. I know of the SSL support for AMI, but it will not protect against a malicious end user.

Is this question too basic, or just cannot be done other than with the proxy 
approach?

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[asterisk-users] How to allow AMI access to Originate yet deny Application: System

[Alex Villací­s Lasso]

While doing a security audit on a system I maintain, I stumbled upon an unvalidated use of a variable to compose an Originate request to the local Asterisk instance via AMI. Taking as an example an earlier exploit for FreePBX, I realized that this, 
combined with Application: System as an injected value, could allow arbitrary code execution. I am in the process of fixing all instances of this bug in our system. However, there are third parties that plug into our system, and that reconfigure the 
manager.conf file to allow remote access to AMI logins that allow Originate (by default, the manager.conf remains configured to deny login to any system except localhost). I want to have a guideline on how to proceed in order to make these applications 
work, without allowing malicious users to compromise the system. I know that one way to proceed is to deny remote access to AMI, and build an application-specific proxy that will perform the Originate on behalf of the remote requester, after filtering the 
values. However, I want to know if there is a simpler way to remove the danger of code execution while allowing applications to use AMI to place calls.


The intended scenario is that a remote desktop application (for Windows) is configured with the AMI credentials, and connects over the LAN to Asterisk in order to place calls and otherwise monitor the system. The attack I want to protect against is that of 
a malicious user that collects the credentials from the desktop application and proceeds to use the Application: System trick. I know of the SSL support for AMI, but it will not protect against a malicious end user.


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Re: [asterisk-users] Error to install Asterisk

[Alex Villací­s Lasso]

El 06/03/13 11:52, Carlos Alvarez escribió:
I'm going to make an observation here that may upset you, and I don't mean it to, but it's fact.  If you are so unfamiliar with Linux, you will have a bad time managing Asterisk servers.  You really need to know how to use the OS before you can learn to 
manage services running on it.  I strongly suggest one of the all-in-one Asterisk variants like AsteriskNOW.  There is simply no way to run a production server without having to do systems management regularly.



On Wed, Mar 6, 2013 at 3:01 AM, termo termosel fermit...@hotmail.com 
mailto:fermit...@hotmail.com wrote:

Hi,

this is the outpu to df -h command:

root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
S.ficherosTam.  Usado Disp. % Uso Montado en
/cow   14G  4,5G  8,7G  34% /
udev  999M  4,0K  999M   1% /dev
tmpfs 403M  860K  402M   1% /run
/dev/sdb1 799M  693M  106M  87% /cdrom
/dev/loop0668M  668M 0 100% /rofs
tmpfs1006M   44K 1006M   1% /tmp
none  5,0M 0  5,0M   0% /run/lock
none 1006M  100K 1006M   1% /run/shm

Jordi


Look carefully at the df -h output. It seems that the OP is trying to install 
and run asterisk from inside an Ubuntu livecd session. Whatever the result of 
the installation, it will be wiped out on the next restart.
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Re: [asterisk-users] Exiting the queue doesn't work

[Alex Kauffmann]

On 3/4/2013 6:27 AM, Gertjan Baarda wrote:

Dear guru's

Hopefully someone can shed some light in my issue. I have created a
queue with a ringall strategy and all works fine. I want a caller to be
able to exit the queue so they can leave a message. I've added the H
option in queue command so callers can press * to exit. So far all well,
on the cli there is a message the caller pressed * and the extensions
stops ringing. But here's the thing: the caller stays in the queue and
after a few seconds the extensions starts to ring again. I want the call
to leave the queue and continue in the dialplan.

After extensively googling the issue, I've found everything (also bug
related), accept my answer. What am I missing here?

It's Asterisk 1.8 on a Debianbox.

Thanks!
Gertjan


--


Look at context= in queues.conf.

Alex


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Re: [asterisk-users] Exiting the queue doesn't work

[Alex Kauffmann]

On 3/4/2013 7:27 AM, Gertjan Baarda wrote:

ok, resumé: When I use the n option in the queue command I can let the
caller exit the queue and send the call to a IVR-ish context and ask if
he wants to leave a message. I can timeout this an then place the call
back in the queue. When I use this approach, what will the new position
be of the caller? Not back in line I hope?


On Mon, Mar 4, 2013 at 2:12 PM, Bharat Lalcheta
bharatlalch...@gmail.com mailto:bharatlalch...@gmail.com wrote:

yes,

context parameter in queue.conf is more likely option for you. It will
work during MOH too.


On Mon, Mar 4, 2013 at 6:33 PM, Bharat Lalcheta
bharatlalch...@gmail.com mailto:bharatlalch...@gmail.com wrote:
  No its again place into queue so its start with new available
position.
 
  However, mostly all users remain in same position if he come again in
  queue using below scenario.
 
  Regards,
 
  Bharat Lalcheta
 
 
 
 
 
  On Mon, Mar 4, 2013 at 6:22 PM, Gertjan Baarda
gertjan.baa...@gmail.com mailto:gertjan.baa...@gmail.com wrote:
  Ah.. thanks! That was the light I needed. When the caller is
placed back in
  the queue, I presume the caller remain it's position in the queue?
 
 
  On Mon, Mar 4, 2013 at 1:45 PM, Bharat Lalcheta
bharatlalch...@gmail.com mailto:bharatlalch...@gmail.com
  wrote:
 
  Hii,
 
  Queue(testq,H) feature works once call connected with agent
i.e. not
  work during MOH.
 
  Also once you disconnect call using H (*) option, it will not
useful
  to leave voicemail.
 
  Instead you use, queue timeout option and ask caller to leave voice
  mail if he wants else put back him to queue again.
 
  Hope it helps you out.
 
  Regards,
 
  Bharat Lalcheta
 
  On Mon, Mar 4, 2013 at 5:57 PM, Gertjan Baarda
gertjan.baa...@gmail.com mailto:gertjan.baa...@gmail.com
  wrote:
   Dear guru's
  
   Hopefully someone can shed some light in my issue. I have
created a
   queue
   with a ringall strategy and all works fine. I want a caller
to be able
   to
   exit the queue so they can leave a message. I've added the H
option in
   queue
   command so callers can press * to exit. So far all well, on
the cli
   there is
   a message the caller pressed * and the extensions stops
ringing. But
   here's
   the thing: the caller stays in the queue and after a few
seconds the
   extensions starts to ring again. I want the call to leave the
queue and
   continue in the dialplan.
  
   After extensively googling the issue, I've found everything
(also bug
   related), accept my answer. What am I missing here?
  
   It's Asterisk 1.8 on a Debianbox.
  
   Thanks!
   Gertjan
  
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  --
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  --
  Bharat Lalcheta



--
Bharat Lalcheta

Why not set context= some context and use periodic announcement to say 
You may press * at any time to leave us a message.  You can also  play 
the message before entering the queue (only once and caller may forget 
what key to press).  This way the caller looses their position in the 
queue only if they choose to leave a message.


Alex




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Re: [asterisk-users] Exiting the queue doesn't work

[Alex Kauffmann]

On 3/4/2013 8:00 AM, Gertjan Baarda wrote:

This will only work with the n option in the queue command and retry=0
in queue.conf. Is it not?

On Mon, Mar 4, 2013 at 2:55 PM, Alex Kauffmann akauf...@prodigy.net.mx
mailto:akauf...@prodigy.net.mx wrote:

On 3/4/2013 7:27 AM, Gertjan Baarda wrote:

ok, resumé: When I use the n option in the queue command I can
let the
caller exit the queue and send the call to a IVR-ish context and
ask if
he wants to leave a message. I can timeout this an then place
the call
back in the queue. When I use this approach, what will the new
position
be of the caller? Not back in line I hope?


On Mon, Mar 4, 2013 at 2:12 PM, Bharat Lalcheta
bharatlalch...@gmail.com mailto:bharatlalch...@gmail.com
mailto:bharatlalcheta@gmail.__com
mailto:bharatlalch...@gmail.com wrote:

 yes,

 context parameter in queue.conf is more likely option for
you. It will
 work during MOH too.


 On Mon, Mar 4, 2013 at 6:33 PM, Bharat Lalcheta
 bharatlalch...@gmail.com mailto:bharatlalch...@gmail.com
mailto:bharatlalcheta@gmail.__com
mailto:bharatlalch...@gmail.com wrote:
   No its again place into queue so its start with new
available
 position.
  
   However, mostly all users remain in same position if he
come again in
   queue using below scenario.
  
   Regards,
  
   Bharat Lalcheta
  
  
  
  
  
   On Mon, Mar 4, 2013 at 6:22 PM, Gertjan Baarda
 gertjan.baa...@gmail.com mailto:gertjan.baa...@gmail.com
mailto:gertjan.baarda@gmail.__com
mailto:gertjan.baa...@gmail.com wrote:
   Ah.. thanks! That was the light I needed. When the
caller is
 placed back in
   the queue, I presume the caller remain it's position in
the queue?
  
  
   On Mon, Mar 4, 2013 at 1:45 PM, Bharat Lalcheta
 bharatlalch...@gmail.com mailto:bharatlalch...@gmail.com
mailto:bharatlalcheta@gmail.__com
mailto:bharatlalch...@gmail.com

   wrote:
  
   Hii,
  
   Queue(testq,H) feature works once call connected with
agent
 i.e. not
   work during MOH.
  
   Also once you disconnect call using H (*) option, it
will not
 useful
   to leave voicemail.
  
   Instead you use, queue timeout option and ask caller
to leave voice
   mail if he wants else put back him to queue again.
  
   Hope it helps you out.
  
   Regards,
  
   Bharat Lalcheta
  
   On Mon, Mar 4, 2013 at 5:57 PM, Gertjan Baarda
 gertjan.baa...@gmail.com mailto:gertjan.baa...@gmail.com
mailto:gertjan.baarda@gmail.__com
mailto:gertjan.baa...@gmail.com

   wrote:
Dear guru's
   
Hopefully someone can shed some light in my issue. I
have
 created a
queue
with a ringall strategy and all works fine. I want a
caller
 to be able
to
exit the queue so they can leave a message. I've
added the H
 option in
queue
command so callers can press * to exit. So far all
well, on
 the cli
there is
a message the caller pressed * and the extensions stops
 ringing. But
here's
the thing: the caller stays in the queue and after a few
 seconds the
extensions starts to ring again. I want the call to
leave the
 queue and
continue in the dialplan.
   
After extensively googling the issue, I've found
everything
 (also bug
related), accept my answer. What am I missing here?
   
It's Asterisk 1.8 on a Debianbox.
   
Thanks!
Gertjan
   
--
   


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Re: [asterisk-users] Point a Digium phone to a configuration URL using mDNS without DPMA or DHCP option 66

[Alex Villací­s Lasso]

El 28/02/13 08:40, A J Stiles escribió:

On Wednesday 27 February 2013, Alex Villací­s Lasso wrote:

I have the following scenario. A small network has DHCP but does not
publish option 66. An Asterisk server is on the network, but the Asterisk
version does not support DPMA and it is hard to switch the version.
However, there is a possibility to have a web server and an mDNS (Avahi)
server. I have been reading about provisioning Digium phones without DPMA,
and it mentions that option 66 can specify the URL prefix to use for the
XML configuration. Is there a way to specify the same through Avahi mDNS?
Something along the lines of this (of course this does not work, but gives
an idea of what I am looking for):

DPMA is proprietary and toxic, so you are quite right not to use it.

If you add a line like
options tftp-server-name 10.0.0.145
to your DHCP server's /etc/dhcp/dhcpd.conf , then this will give you the
option 66 you need. And note that even although it's advertised as a TFTP
server, the Digium phones actually put out a HTTP request aot a TFTP request
(that took some head-scratching to figure out).  So you will need to have
Apache running on 10.0.0.145; and a bunch of configuration files named like
000fd300.cfg
where the digits of the filename are the MAC address of the phone, in the web
server's default document root  (/var/www/ on Debian systems).

I forgot to point out that in my scenario, I may not have any administrative control over the DHCP server. Otherwise, the way you indicate would be obvious. Anyway, I received an email saying that mDNS cannot support pointing a Digium phone to anything 
other than a DPMA-enabled Asterisk.


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[asterisk-users] Point a Digium phone to a configuration URL using mDNS without DPMA or DHCP option 66

[Alex Villací­s Lasso]

I have the following scenario. A small network has DHCP but does not publish option 66. An Asterisk server is on the network, but the Asterisk version does not support DPMA and it is hard to switch the version. However, there is a possibility to have a web 
server and an mDNS (Avahi) server. I have been reading about provisioning Digium phones without DPMA, and it mentions that option 66 can specify the URL prefix to use for the XML configuration. Is there a way to specify the same through Avahi mDNS? 
Something along the lines of this (of course this does not work, but gives an idea of what I am looking for):


?xml version=1.0 standalone='no'?
!DOCTYPE service-group SYSTEM avahi-service.dtd
service-group
nameDigium Phone Configuration Server/name
service
type_digiumproxy._udp/type
port443/port
txt-recordconfig_server_url=https://10.0.0.145/provisioning_scripts/http_provisioning.php?authhash=Xamp;filename=/txt-record
txt-recordserviceType=configuration/txt-record
/service
/service-group

I want the phone to request 
https://10.0.0.145/provisioning_scripts/http_provisioning.php?authhash=Xamp;filename=/001915abcdef.cfg;.

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Re: [asterisk-users] Asterisk AMI - Create a daemon (background process)

[Alex Villací­s Lasso]

El 24/02/13 07:30, Shahid H escribió:

I wanted to create a daemon (background process) in PHP. A daemon will use 
socket to connect with Asterisk AMI to send events and listen the actions.

A daemon will also listen the commands from agents via HTTP, for example:  A 
agent pressed a hang up button on a browser - it will send http command to a 
daemon.  A daemon received a command and will then send Hang Up Action to AMI.

How should a daemon process be designed to listen multiple actions and events? 
For example: 50 agents currently on the calls and how should a daemon to 
monitor the Actions/Events from 50 agents?


You may want to take a look to the Elastix CallCenter code. This project has a 
daemon component to implement the autodialer, written in PHP. This code already 
solves the AMI connection issue and the multiple agent issue.

http://elastix.svn.sourceforge.net/viewvc/elastix/trunk/apps/extras/callcenter/

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Re: [asterisk-users] PRI can receive calls but cannot dial out

[Alex Kauffmann]

]
PRI Span: 4  Calling Party Number (len= 8) [ Ext:

0  TON: National Number (2)  NPI: ISDN/Telephony
Numbering Plan (E.164/E.163) (1)

PRI Span: 4 


Presentation: Presentation
allowed, User-provided, not screened (0)  '4053' ]

PRI Span: 4  [70 0a 80 36 35 36 36 36 30 34 39 39]
PRI Span: 4  Called Party Number (len=12) [ Ext:

1  TON: Unknown Number Type (0)  NPI: Unknown
Number Plan (0)  'XX' ]

PRI Span: 4 q931.c:6291 q931_setup: Call 32774 enters

state 1 (Call Initiated).  Hold state: Idle

   -- Called DAHDI/g2/XX
PRI Span: 4 T303 timed out.  cref:32774
PRI Span: 4
PRI Span: 4  DL-DATA request
PRI Span: 4  Protocol Discriminator: Q.931

(8)  len=32

PRI Span: 4  TEI=0 Call Ref: len= 1 (reference

6/0x6) (Sent from originator)

PRI Span: 4  Message Type: SETUP (5)
PRI Span: 4 TEI=0 Transmitting N(S)=11, window is open

V(A)=11 K=1

PRI Span: 4
PRI Span: 4  Protocol Discriminator: Q.931

(8)  len=32

PRI Span: 4  TEI=0 Call Ref: len= 1 (reference

6/0x6) (Sent from originator)

PRI Span: 4  Message Type: SETUP (5)
PRI Span: 4  [04 03 80 90 a3]
PRI Span: 4  Bearer Capability (len= 5) [ Ext:

1  Coding-Std: 0  Info transfer capability: Speech
(0)

PRI Span: 4 


 Ext: 1  Trans mode/rate: 64kbps,
circuit-mode (16)

PRI Span: 4 


   User information layer 1: A-Law (35)

PRI Span: 4  [18 01 81]
PRI Span: 4  Channel ID (len= 3) [ Ext: 1

IntID: Implicit  BRI  Spare: 0
Preferred  Dchan: 0

PRI Span: 4 


ChanSel: B1 channel

PRI Span: 4 

]

PRI Span: 4  [6c 06 21 80 34 30 35 33]
PRI Span: 4  Calling Party Number (len= 8) [ Ext:

0  TON: National Number (2)  NPI: ISDN/Telephony
Numbering Plan (E.164/E.163) (1)

PRI Span: 4 


Presentation: Presentation
allowed, User-provided, not screened (0)  '4053' ]

PRI Span: 4  [70 0a 80 36 35 36 36 36 30 34 39 39]
PRI Span: 4  Called Party Number (len=12) [ Ext:

1  TON: Unknown Number Type (0)  NPI: Unknown
Number Plan (0)  'XX' ]

PRI Span: 4 T303 timed out.  cref:32774
PRI Span: 4 q931.c:6180 t303_expiry: Call 32774 enters

state 0 (Null).  Hold state: Idle

PRI Span: 4 Fake clearing.  cref:32774
PRI Span: 4 q931.c:9551 pri_internal_clear: alive 1,

hangupack 1

Span 4: Processing event PRI_EVENT_HANGUP(6)
   -- Span 4: Channel 0/1 got hangup,

cause 18

PRI Span: 4 q931.c:7092 q931_hangup: Hangup other

cref:32774

PRI Span: 4 q931.c:6849 __q931_hangup: ourstate Null,

peerstate Null, hold-state Idle

PRI Span: 4 Destroying call 0xb85c61d0, ourstate Null,

peerstate Null, hold-state Idle

   -- Hungup 'DAHDI/i4/XX-6'
 == Everyone is busy/congested at this time

(1:0/0/1)

   -- Auto fallthrough, channel

'SIP/4053-0003' status is 'CHANUNAVAIL'

   -- Executing [h@company:3]

Hangup(SIP/4053-0003, ) in new stack

 == Spawn extension (company, h, 3) exited

non-zero on 'SIP/4053-0003'


Note that incoming calls via this PRI work correctly.

Asterisk 11.0.1
latest libpri and dahdi.

Thanks,

Vieri


--- On Wed, 12/5/12, Thorsten Göllner t...@ovm-group.com wrote:


From: Thorsten Göllner t...@ovm-group.com
Subject: Re: [asterisk-users] PRI can receive calls but cannot dial out
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Cc: Vieri rentor...@yahoo.com
Date: Wednesday, December 5, 2012, 8:21 AM
Hi!

1) How long does the outdial take? Does the Dial-Command
return immediatly?

2) Maybe dial-out is blocked by your carrier? Did you try to
open a
trouble ticket there?

3) What number do you try to call? Did you try some
different number?
Alway the same problem?

You receive ISDN-Cause-Code 18. Not sure though, but I would
open a
troubke ticket at your carrier.

-Thorsten-


Hi,

1) just a few seconds (max. 5)
2) it isn't blocked by the carrier (see below)
3) always the same problem regardless of dialed number

I did another test and noticed that if the first port of the card is 
disconnected (no cable) then calls can't go out (they can only come in). 
However, if I connect the first port then calls do go out. Why's that?
Could there be something wrong with my /etc/dahdi/system.conf or 
chan_dahdi.conf (see above)?

Thanks,

Vieri



The first port has channels defined in group 2, but the port is down. 
Have you tried dialing out with G2 as opposed to g2?


Alex

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Re: [asterisk-users] Change phone display from queue calls

[Alex Kauffmann]

On 12/6/2012 12:32 PM, Carlos Alvarez wrote:

We are trying to set up a system where the calls from the queue show a
specific name or number on the phone.  The calls would come into one of
a few dozen DID numbers, each one for a specific company.  The agent
needs to know which company the call is for and answer appropriately.
  I've done a lot of this in dialplans but haven't found a way to do it
in a queue.

--
Carlos Alvarez
TelEvolve
602-889-3003



We either set callerid(name) with a fixed name depending on the trunk, 
or set it from a variable we get from a database based on 
callerid(number) before we dial the queue.


The new value for callerid(name) will show on the agent's screen. 
Setting Callerid(number) will work as well.


alex


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[asterisk-users] Query list of defined channel variables via AMI

[Alex Villací­s Lasso]

Is there a way to list the names of the channel variables that are currently 
defined on a given channel via AMI? I know of GetVar and SetVar, but GetVar 
needs the name of the variable to get.

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Re: [asterisk-users] leading ghost 0

[Alex Kauffmann]

On 11/21/2012 10:53 AM, gincantalupo wrote:

Alex,

I had already tried itreloading chan_dahdi.so module is enough...I
saw Asterisk was behaving differently after reload. To tell the truth,
setting pridialplan=unknown causes Asterisk to stop reading following
channels configuration...it says pridialplan is already unknown so it
stops evaluating chan_dahdi.conf file useless to say that all n+1
channels do not work. Maybe it is a bug but with that parameter set in
that way I cannot dial.

I'm sure Asterisk is dialling the right number:

[2012-11-21 09:05:29] VERBOSE[8314] logger.c:  [70 0b a1 33 34 39 3x 3x
3x 3x 3x 3x 34]
[2012-11-21 09:05:29] VERBOSE[8314] logger.c:  Called Number (len=13) [
Ext: 1  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan
(E.164/E.163) (1)  '3497078884' ]
[2012-11-21 09:05:29] VERBOSE[8314] logger.c: q931.c:3134 q931_setup:
call 32781 on channel 6 enters state 1 (Call Initiated)
[2012-11-21 09:05:29] VERBOSE[8314] logger.c: -- Called 6/349xx4

I'm starting to think it is a telco problem... in case I'd change some
parameter like pridialplan or similar, shouldn't I just see a leading 0
in the frame like this:
[70 0b a1 *30* 33 34 39 3x 3x 3x 3x 3x 3x 34] added by Asterisk/DAHDI??

I've used this page as reference about frame fields:
http://www.acacia-net.com/wwwcla/protocol/q931_ie.htm

Thank you.

Giorgio Incantalupo


On 11/20/2012 05:23 PM, Alex Kauffmann wrote:

On 11/20/2012 8:03 AM, gincantalupo wrote:

Hi Leandro,

I'm sure nobody has added something... tried prilocaldialplan and
pridialplan but nothing changed.
Question: if pridialplan or prilocaldialplan would work, should I see
the 0 inside PRI frame with intense debug or it is hidden?

Yes...the technician did it...there is only one cable.

Maybe it is the socket circuitry that has something wrong but I do not
know ho to check.

Asap I'll be on site I'll do more testing.

Thank you

Giorgio

On 11/20/2012 01:13 PM, Leandro Dardini wrote:

That is a real mistery! I like a lots these cases when all seems not
working despite all being correctly configured, but you know first or
later you'll find the answer.

From your website, it seems you are selling/renting PBX based on
asterisk, so you can be sure nobody has messed with the asterisk or
dahdi source code adding a zero... I am sure you have already tried
with a brand new server.

Have you checked the pridialplan and prilocaldialplan setting?

If I was in your shoes, I'll get another server, with a PRI configured
as master and hook it at your PBX to really check if the zero is sent.

Does the technician try to make phone calls from the same network
cable you are using?

Leandro


2012/11/20 gincantalupo gincantal...@fgasoftware.com
mailto:gincantal...@fgasoftware.com

Hi Leandro,

thanks for your answer.

I already have tried those parameters but without any positive
result.

The telco technician has tried the line with its machine and it
worked...remote telco technicians say they get a leading zero...
I'm thinking there is something strange in the middle that adds
the zero but do not know what it is.
Strange is the fact that you can call some numbers with or without
the prefix zero...
Moreover we had no problem with the previous telco (fastweb).

So we can only call PTSN numbersnot mobile phones.

Giorgio


On 11/20/2012 11:12 AM, Leandro Dardini wrote:

2012/11/20 gincantalupo gincantal...@fgasoftware.com
mailto:gincantal...@fgasoftware.com

Hi all,

I have problems dialling out because my new telco (the
previous gave no problems) tells me my PBX adds a leading 0
and that's why I cannot dial out (but I can receive calls).

I make a small extensions.conf as a test:

exten = 666,1,Dial(DAHDI/g1/339xx)
but cannot dial out

Curious thing is that
exten = 666,1,Dial(DAHDI/g1/0233xx)
and
exten = 666,1,Dial(DAHDI/g1/233xx)
call the same number!!!

Line in use is a PRI.

My Asterisk version is 1.4.26.2
dahdi version: 2.2.0.2
wanpipe-3.4.6

I checked with intense pri debug and see no 0 inside
frames

How can I really be SURE Asterisk is not adding some leading
zero?

Thank you.

Giorgio.


I have never heard of a way to automatically add digits when
using PRI, however can you check your chan_dahdi.conf about the
following lines:

internationalprefix =
nationalprefix =
localprefix =

If presents, try messing with them. If you are using the PRI in
Italy, every provider has PRI configured in its own way, some
time even the same provider is configuring PRI lines in multiple
times, but often the problems are on receiving the calls (like
calls with and without the area code, with or without the leading
zero, etc. etc.)

Leandro


--


The prilocaldialplan parameter

Re: [asterisk-users] leading ghost 0

[Alex Kauffmann]

On 11/20/2012 8:03 AM, gincantalupo wrote:

Hi Leandro,

I'm sure nobody has added something... tried prilocaldialplan and
pridialplan but nothing changed.
Question: if pridialplan or prilocaldialplan would work, should I see
the 0 inside PRI frame with intense debug or it is hidden?

Yes...the technician did it...there is only one cable.

Maybe it is the socket circuitry that has something wrong but I do not
know ho to check.

Asap I'll be on site I'll do more testing.

Thank you

Giorgio

On 11/20/2012 01:13 PM, Leandro Dardini wrote:

That is a real mistery! I like a lots these cases when all seems not
working despite all being correctly configured, but you know first or
later you'll find the answer.

From your website, it seems you are selling/renting PBX based on
asterisk, so you can be sure nobody has messed with the asterisk or
dahdi source code adding a zero... I am sure you have already tried
with a brand new server.

Have you checked the pridialplan and prilocaldialplan setting?

If I was in your shoes, I'll get another server, with a PRI configured
as master and hook it at your PBX to really check if the zero is sent.

Does the technician try to make phone calls from the same network
cable you are using?

Leandro


2012/11/20 gincantalupo gincantal...@fgasoftware.com
mailto:gincantal...@fgasoftware.com

Hi Leandro,

thanks for your answer.

I already have tried those parameters but without any positive result.

The telco technician has tried the line with its machine and it
worked...remote telco technicians say they get a leading zero...
I'm thinking there is something strange in the middle that adds
the zero but do not know what it is.
Strange is the fact that you can call some numbers with or without
the prefix zero...
Moreover we had no problem with the previous telco (fastweb).

So we can only call PTSN numbersnot mobile phones.

Giorgio


On 11/20/2012 11:12 AM, Leandro Dardini wrote:

2012/11/20 gincantalupo gincantal...@fgasoftware.com
mailto:gincantal...@fgasoftware.com

Hi all,

I have problems dialling out because my new telco (the
previous gave no problems) tells me my PBX adds a leading 0
and that's why I cannot dial out (but I can receive calls).

I make a small extensions.conf as a test:

exten = 666,1,Dial(DAHDI/g1/339xx)
but cannot dial out

Curious thing is that
exten = 666,1,Dial(DAHDI/g1/0233xx)
and
exten = 666,1,Dial(DAHDI/g1/233xx)
call the same number!!!

Line in use is a PRI.

My Asterisk version is 1.4.26.2
dahdi version: 2.2.0.2
wanpipe-3.4.6

I checked with intense pri debug and see no 0 inside frames

How can I really be SURE Asterisk is not adding some leading
zero?

Thank you.

Giorgio.


I have never heard of a way to automatically add digits when
using PRI, however can you check your chan_dahdi.conf about the
following lines:

internationalprefix =
nationalprefix =
localprefix =

If presents, try messing with them. If you are using the PRI in
Italy, every provider has PRI configured in its own way, some
time even the same provider is configuring PRI lines in multiple
times, but often the problems are on receiving the calls (like
calls with and without the area code, with or without the leading
zero, etc. etc.)

Leandro


--


The prilocaldialplan parameter is for inbound so you would have seen no 
changes.  Did you try:


pridialplan=unknown

Did you restart dahdi and asterisk after the changes?

Alex


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Re: [asterisk-users] DAHDI and Tiger320 Chip

[Alex Villací­s Lasso]

El 25/10/12 07:46, Antonio Modesto escribió:

Hi,

I've got a ISDN Interface: /Tiger Jet/ Network Inc. Tiger3XX Modem/ISDN interface, I'm trying to use it with DAHDI 2.6 but it doesn't work, I'm thinking that dahdi doesn't support this device, I've loaded all of available dahdi drivers and none of them 
worked. Does anybody know what I can do to use it with DAHDI/Asterisk? I load dahdi and the modules, but it doesn't recognize any channel under /dev/dahdi/*




What does lspci -vnnn say about the device?
What about running dahdi_hardware ?
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Re: [asterisk-users] How to tie orders taken to specific CDR records

[Alex Kauffmann]

On 10/25/2012 11:18 AM, Mitch Claborn wrote:

Our phone operators work off of an Asterisk queue.  They take calls from
customers and take orders with our back end systems.  What I need to be
able to do is tie the orders taken to the specific CDR record that
reflects the call from which the order originated.

The typical/sample CDR table doesn't have a primary key.  I can add an
auto-generated PK, but the CDR is not written until the call ends, when
the orders have already been placed.  (Even if the CDR was written
earlier, could I retrieve the generated PK from it in the dialplan
somehow?)

Is there some combination of fields in the CDR that might uniquely
identify a specific call?

Open to any and all ideas.


Try looking at the queue_log.  Configure your system to log to mysql and 
you should be able to get everything you need in realtime.


Alex

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Re: [asterisk-users] Agents in more than one queue at once

[Alex Forster]

 In general there is no guaarantee as which call will connect; each queue is
 independent AFAIK.

Lenz- big fan :) And I'm sure this topic is of interest to you...

I'll admit, I had a feeling that it's random would be the response to my
original question. I remember reading the app_queue code a while back and
getting the impression that the logic was something like-

* Loop through the list of all queued calls in the system, one by one. If the
current call is 'next' in its respective queue (as defined by the queue
scheduling algo), then we need to find an agent to take this call...
* Loop through the list of agents who are members of this queue. If the current
agent is available to take a call, then send the call to the available agent.

This logic leaves the above mentioned oversight where at no point is it ever
considered that an agent may be a member of more than one queue. Even with a
fair scheduling algo applied to each queue, this bug causes queues with large
numbers of waiting calls and/or large numbers of available agents to starve
other queues.

Effectively, this bug makes skills-based routing impossible, because unique
skills can not exist. Additionally, regardless of how many callers are in
queue, this bug causes hold times to increase - significantly, in our case.

This is an important oversight in my opinion because it is the only way that
skills-based routing can be implemented within the framework of app_queue.
Without fixing this issue, Asterisk can not claim to have a reliable method
of implementing skills-based routing.


*DEVELOPERS* - If I took a crack at fixing this issue, what general tips do
you have for me to make it most likely that my solution can be integrated
into HEAD? I believe I can justify spending some time at work to deal with
this, but not without at least a decent chance that the work will be
integrated into mainline (assuming it doesn't suck, of course :)

Alex Forster


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[asterisk-users] Tell apart between network disruption and asterisk restart via AMI

[Alex Villací­s Lasso]

I have a program that connects to the Asterisk Manager Interface through port 5038 on a remote machine. Suppose I get a TCP disconnection on my program. The program will then attempt to reconnect to the AMI and will eventually succeed. Is there a way to 
check whether the disconnection was caused by a network disruption, or an Astersk restart/crash? In other words, is the Asterisk process I contacted now the same as the one I was connected before, or is it a different one? The reason I want to know is that 
I have a cache of information that is costly to parse (scales linearly with the number of extensions) and I want to know how to realize that the information is now stale.


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Re: [asterisk-users] Agents in more than one queue at once

[Alex Forster]

Are there any developers that are familiar with the Queue() app implementation
and how it distributes calls?

Alex Forster


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[asterisk-users] Agents in more than one queue at once

[Alex Forster]

My company has been running Asterisk 1.6.2.19-1_centos5 from the official
yum repo, and for a while now I've been receiving complaints from our call
centers about calls not being routed in the most efficient order.

I'll explain with a simplified scenario--

Let's say I have two queues: A and B. I have one agent, Alice, who is a
member of both of these queues. While Alice is busy on a call, one person
calls in to queue A, and then, several moments later, another person calls
in to queue B.

At this point, note that both callers waiting on hold are position 1 in
their respective queues. A queue show might look like this...

 A has 1 calls (max unlimited) in 'leastrecent' strategy (0s
holdtime, 533s talktime), W:1, C:1, A:0, SL:100.0% within 60s
Members:
   21 (Local/21@from-queue/n) (dynamic) (In use) has taken 1 calls
(last was 533 secs ago)
Callers:
   1. SIP/Trunk-eb17 (wait: 1:14, prio: 0)

 B has 1 calls (max unlimited) in 'leastrecent' strategy (0s
holdtime, 533s talktime), W:1, C:1, A:0, SL:100.0% within 60s
Members:
   21 (Local/21@from-queue/n) (dynamic) (In use) has taken 1 calls
(last was 533 secs ago)
Callers:
   1. SIP/Trunk-eb1e (wait: 0:45, prio: 0)

My question is: when Alice gets off the phone, which call will she get? My
expectation is that she will get the call which has been waiting longer,
but I'm not sure that's actually the case.

Alex Forster
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Re: [asterisk-users] How to log caller IP address in the CDR?

[Alex Oniciuc]

Ishfaq is right, that's the way to go.
Here's a dialplan line to help you achieve that:

exten = YOUREXTEN_CHANGE_ME,PRIORITY_CHANGE_ME,Set(CDR(UserField)=SIP
HEADER CONTACT: ${SIP_HEADER(CONTACT)}, SIPURI: ${SIPURI}, SIP PEER IP:
${SIPCHANINFO(peerip)}, SIP RECEIVED IP: ${SIPCHANINFO(recvip)}, SIP FROM:
${SIPCHANINFO(from)})

Alex

2012/10/5 Ishfaq Malik i...@pack-net.co.uk

 On Fri, 2012-10-05 at 14:10 +0200, Benoit Panizzon wrote:
  Hello
 
  We had this situation:
 
  Some bot-net did try to guess SIP logins and finally succeeded. The
 Asterisk
  Server was abused to call a large number of expensive destinations.
 
  It is clear that the sip logins have been passed to various persons
 (probably
  posted on a forum somewhere inviting to do 'free calls').
 
  Right after the affected password was changed, the message log shows
 which IP
  did try to make calls.
  We also got a few snapshots of 'sip show channels' which show the ip
 addresses
  of active in call connections.
  So basicly it is known, who abused the service. It was abused from
 multiple IP
  addresses at the same time.
 
  Legal steps against the abusers have been taken, but to claim the costs
 of the
  damage they generated we would need to know exactly which calls
 originated
  from which IP address to put an exact sum of damage done by each of the
  abusers.
 
  Well for this case it is too late now. But is there a way to get the IP
  Address of the SIP Client being logged in each CDR?
 
  Kind regards
 
  Benoit Panizzon
 
  --

 Hi

 Get info using function SIPCHANINFO
 https://wiki.asterisk.org/wiki/display/AST/Function_SIPCHANINFO

 Set it to CDR using CDR(userfield)

 The above are for 1.8

 Regards

 Ish

 --
 Ishfaq Malik i...@pack-net.co.uk
 Department: VOIP Support
 Company: Packnet Limited
 t: +44 (0)845 004 4994
 f: +44 (0)161 660 9825
 e: i...@pack-net.co.uk
 w: http://www.pack-net.co.uk

 Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
 NORTH, MANCHESTER
 SCIENCE PARK, MANCHESTER, M156SE
 COMPANY REG NO. 04920552


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Re: [asterisk-users] So long, and thanks for all the fish!

[Alex Oniciuc]

No Kevin, we thank you for the fish, for sharing the knowledge and for
having the patience...
Good luck on your new journey!

P.S. As a token of our appreciation, just a word and we'll make the life of
the new Director of Software Technologies miserable!


2012/7/31 Kevin P. Fleming kpflem...@digium.com

 I've been with Digium for just over seven years, and it's been an
 incredible experience that I wouldn't have traded for anything. When
 Mark Spencer invited me to visit Digium (and Huntsville) in early
 2005, I could not have dreamed that I'd end up working for such an
 exciting, innovative company, finding a wife, and meeting hundreds of
 people (many of whom are now friends) around the world. It's been a
 time of tremendous personal and career growth, and my wonderful
 colleagues at Digium and in the Asterisk open source community have
 been directly responsible for most of that.

 Recently, though, I've been presented an opportunity to take on a new
 challenge and this has resulted in my acceptance of a new job, in a
 new industry. In the middle of September, I'll start working for
 Bloomberg, L.P., in the Office of the CTO, helping to lead their
 nascent open source initiative. I'll be working to bring the power of
 open source software, open standards, and community building to the
 financial market data services industry, where it is sorely needed
 (and overdue). Michelle and I will be relocating to the greater New
 York City area, but Michelle will continue in her role as Digium's
 in-house counsel. Because of our need to relocate, I'll only be at
 Digium until August 8th, although I'll be in Huntsville until around
 Labor Day.

 This is yet another incredibly exciting, career changing opportunity
 in my life, and I can't wait to see what it will bring. I'll be
 forever thankful for the opportunity that Digium and the Asterisk
 community provided me to learn, grow and find the place where my skills
 and experience are the most valuable (to both myself and my employer).

 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?

[Alex Balashov]

SIPAddHeader() comes to mind. :-) 




-- Alex

--
Sent from my Samsung mobile, and thus lacking in the refinement one might 
expect from a proper keyboard. 

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/Patrick Lists asterisk-l...@puzzled.xs4all.nl 
wrote:Hi,

The flowroute website mentions that they set callerid on outbound calls 
based on the presence of (in order of preference): 
P-Asserted-Identity, Remote-Party-ID or From:.

I've been trying to make outbound callerid work via flowroute to no 
avail. Does anyone have an extensions.conf / sip.conf snippet howto make 
this work? This is for Asterisk 1.4.44.

Thanks!
Patrick

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Re: [asterisk-users] Elastix 2.3.0.1

[Alex Villací­s Lasso]

El 05/07/12 02:19, Satria Anamarta escribió:

Greetings,

I know this is not a Elastix mailing list, but could anybody please tell where 
I can download Elastix 2.3.0.1 (the latest version) ?

There is only version 2.3.0 (April 2012) on Elastix website, not the 2.3.0.1 
(May 2012), but the changelog information are there.


You can always try installing 2.3.0 then updating everything with yum update.

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Re: [asterisk-users] 2GB Elastix memory limit

[Alex Villací­s Lasso]

El 28/06/12 03:58, resea...@businesstz.com escribió:

I have sevaral elastix installed but all of them show the physical memory
is 2GB while the server has 4GB and some has 8GB. I've upgraded to PAE
kernel but yet i cant see mem beyond 2GB. How can i configure the centos
kernel to use more memory as the server is multipurpose


I think you should use the Elastix mailing lists for this question. But you 
should try using the 64-bit Elastix instead.

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Re: [asterisk-users] Need queue name in CDR

[Alex Ramirez]

  
  
Hi there,

 You can do that by making a context in your dial-plan with the
name of your queue. Then in your queue config put :
context="context you write in your dial-plan".

 That way the asterisk will save the name of your queue in the
'dcontext' column 

 Example:


 ; Context in dialplan
 [in_test]

 ; In queues.conf or in db(if you have it on Real Time)

 [in_test]

 context = in_test


 I hope that's what you are looking for.



El 06/13/2012 03:03 AM, Lenz Emilitri escribi:
It would likely be easier for you to use a tool that
  already processes queue_log information. There are a number
  available :)
  
  2012/6/13 Pratik Shrestha pratik...@gmail.com
Dear All,
  
  
  I am making asterisk report using CDR values given by
asterisk.
  
  
  
I have queues which consist of multiple members (extension).
Also, an extension may be in multiple queues. So, I want CDR
to record the name/number of queue from which the call was
originated.

  
  
  -- 
  Loway - home of QueueMetrics - http://queuemetrics.com
  
  Test-drive WombatDialer alpha @ http://wombatdialer.com
  
  
  
  
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Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.

[Alex Balashov]

Are you certain that this wouldn't be an issue if the phones had low 
re-registration intervals?  Historically, I've seen the Asterisk registrar 
faceplant with throughput in excess of 5-7 registrations/sec, though I have no 
idea as to whether that holds true of newer releases.

--
This message was painstakingly thumbed out on my mobile, so apologies for 
brevity and errors.

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

On May 11, 2012, at 10:40 PM, Kevin P. Fleming kpflem...@digium.com wrote:

 On 05/06/2012 01:39 PM, Paul Belanger wrote:
 
 800 SIP phones on one server? I wouldn't want to do it. Add a SIP proxy
 to your design and have it handle all your SIP.  Then you can load
 balance across multiple asterisk boxes.  You'll be thankful you did this
 at the start, as it will allow you to increase resources more easily.
 
 As has already been pointed out by others in this thread, 800 phones on a 
 single Asterisk server (using Asterisk 1.8.x or later and a decent spec 
 server) is really no problem. If all of those phones are going to be 
 subscribing to hints for a dozen or more of the other phones, then yes, that 
 could be an issue, as the amount of NOTIFY traffic would be quite high... but 
 for registration and normal calling, even if all these phones were in use at 
 once, I would not expect any issues at all due to performance.
 
 The other comments about being able to take down a server for maintenance and 
 not lose calling ability are certainly worth taking into consideration as 
 well, but if your planned deployment would allow for reasonable scheduled 
 maintenance windows, even that wouldn't justify the complexity of adding in 
 one SIP proxy (or a pair of them) to the equation.
 
 -- 
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org
 
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Re: [asterisk-users] Flashphoner

[Alex Balashov]

Really?  Me?

Oh Pavel! I would be inestimably honoured.

On 04/27/2012 01:55 AM, Pavel Ismailov wrote:


Hello!

My name is Pavel Ismailov
and I`m CEO of www.flashphoner.com project.

We noticed that you quite active in Asterisk-user
mail list, and would like to offer you buy signature
in your messages for some monthly price.

Is it interested for you?

--
Thanks,
Pavel Ismailov
skype: pavel.ismailov
www.flashphoner.com




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Re: [asterisk-users] Flashphoner

[Alex Balashov]

On 04/27/2012 01:24 PM, shayne.al...@gmail.com wrote:


congratulations @};-


It's a match made in Heaven.  I have spare signature space to sell, and 
Pavel wants signature space to rent!  At a low introductory rate of 
US$1800/word, he and I are going to make this happen...


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Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Re: [asterisk-users] Flashphoner

[Alex Balashov]

Only the premium dailing minties. The regular flashphoner ones are indebted to 
a complex vanilla ice cream + pork belly + cardboard mixture...

--
This message was painstakingly thumbed out on my mobile, so apologies for 
brevity and errors.

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

On Apr 27, 2012, at 2:43 PM, Jason Parker jpar...@digium.com wrote:

 On 04/27/2012 01:39 PM, Don Kelly wrote:
 What flavor are flashphoner minties?
 
 --Don
 
 
 Dailing flavored.  What else?
 
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Re: [asterisk-users] Monitoring voice-quality with sip/rtp/rtcp

[Alex Balashov]

OpenVZ is not really virtualisation, though for some reason people insist on 
throwing it into the same discursive space as Xen, VMware, HyperV, etc.

--
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Evariste Systems LLC 
235 E Ponce de Leon Ave 
Suite 106
Decatur, GA 30030 
Tel: +1-678-954-0670 
Fax: +1-404-961-1892 
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

Johan Wilfer li...@jttech.se wrote:

2012-04-09 20:22, Carlos Alvarez skrev:
 On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI
 ad...@tootai.net mailto:ad...@tootai.net wrote:


 At first, if your Asterisk is in a VM install it on the real
 server, it solved us on some installations.


 We've gone away from VMs altogether.

I use openVZ to run multiple asterisks on the same server. This works
well and has done for some time. But currently once a week for about
10-15 minutes calls sound like packetloss/jitter occurs. But a week of
traffic captures is heavy... So I need to automate this.

  

 To monitor the traffic, you can use voipmonitor.org
 http://voipmonitor.org


 We purchased the commercial version with a GUI and will tell you that
 the cost/benefit is very clear.  Great tool, pretty cheap ($1k I
 think).  Responsive support.

Sounds very reasonable. Do you run this on a dedicated server, and
configured the switch to duplicate the traffic to the quality server? Or
do you run this on the same server as asterisk?

Thanks for the suggestions!

-- 
Johan Wilfer email: jo...@jttech.se
JT Tech | Developer  webb: http://jttech.se


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Re: [asterisk-users] Asterisk 1.8 and DeadAGI

[Alex Balashov]

Look up the definition of NoOp.  A moral and practical ambivalence inheres in 
that definition.  It is neither more nor less beneficial to use or not use it, 
for it is a NoOp.

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Fax: +1-404-961-1892 
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

bilal ghayyad bilmar...@yahoo.com wrote:

Dears;

In asterisk 1.8, it is not more possible to use DeadAGI?

Also, I found the below commands in the a2billing and I would to ask why it 
set the sequence 1 for the Hangup()? Maybe because it is related to the NoOp? 
How?

[a2billing-callingcard]
exten = _X.,1,NoOp(A2Billing Start)
exten = _X.,n,Answer()
exten = _X.,n,Wait(2)
exten = _X.,n,DeadAgi(a2billing.php,1)
exten = _X.,1,Hangup()

From the other hand, what is the benifit of using NoOp here? Because I tried 
it without NoOp and it was working? 

Regards
Bilal

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Re: [asterisk-users] Rate sheet normalization

[Alex Balashov]

We solve this problem for our customers all the time, in various 
situationally-specific ways. But yes, we are not really in a position to 
genericise it and give it away.  It's not because we are greedy.  The time and 
resources just aren't there.

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A E [Gmail] all.efor...@gmail.com wrote:

On Mon, Mar 12, 2012 at 6:52 PM, Markus unive...@truemetal.org wrote:

 Hi,

 this question is not Asterisk specific, but since there are so many
 experts present on this list, maybe its OK to ask anyways.

 I'm having a hard time normalizing rate sheets from different providers.
 What I mean with this: the goal is to always get the cheapest rate for a
 given destination. What I would like to do is throw like 10 rate sheets
 from different providers together and as output get a single rate sheet
 with only the cheapest rates. However, some providers are listing a
 country, lets say Germany, as code 49 with a specific rate, and another
 provider will list each city individually, and each code separately, e.g.
 Berlin 4930, Hamburg 4940 etc., and probably different cities have
 different rates as well. Now, if the 49 route of the first provider is
 cheaper, my system (a2billing) will still use the more expensive 4930
 code because it is more specific.

 I'm looking for some awesome, smart tool that will automatically
 normalize all these code differences and output a clean ratesheet with
 only the cheapest rates.

 Does such a thing exist? I wonder how everyone else is normalizing their
 different rate sheets. With a homebrewn script?

 Thanks!


Markus,

you're not the first person and certainly not the last person who's ever
asked about this. I had tried this on several mailing lists a little while
ago.  A tool that could handle 10 or maybe even 5 provider rate-sheets all
of which can potentially completely differ in formats from each other. Even
worse are the rate update sheets from each provider which are many a times
different from the initial rate sheets that the provider may have given you
and then again they will differ from the rate updates from the remaining 4
providers you've just painstakingly inserted into your DB.

Given the popularity of Asterisk and other popular OSS based telephony
platforms with several successful businesses running 100s of millions of
minutes, you'd think at least a few have sorted this problem out. But I
believe those who have, never respond to these emails as it took them quite
a bit of effort to create such a tool and aren't willing to just give it
away.

Just what I have observed (and was even blatantly told by someone on some
mailing list, can't remember exactly)

You may have to advertise in the commercial / business list or offer a
bounty. There are several commercial solutions available but I think they
all come as a feature of a larger billing/rating/routing platform

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Re: [asterisk-users] Rate sheet normalization

[Alex Balashov]

On 03/28/2012 03:15 PM, Raj Mathur (राज माथुर) wrote:


Times change -- the way to deal with that is to adapt


I don't think you'll get any serious disagreement on that from anyone 
here.


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Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Re: [asterisk-users] Routing premature media to the calling channel

[Alex Balashov]

I assume you have ruled out NAT and firewall issues?

Between those two, 99% of the reasons why something may not be routed somewhere 
correctly are accounted for. 

If you don't know, your best bet is to take a packet capture or SIP debug on 
the Asterisk server and find out where that early media is going.

--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0671
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

Leandro Dardini ldard...@gmail.com wrote:

Hello,
I have a problem with premature media and inband progress audio. I am using
the latest 1.8.10.1 and this is the setup:

soft phone --- asterisk --- SIP provider

The number I call is giving back some hints via inband audio I am not able
to ear from the soft phone. They stop on the asterisk and are not routed
down the path to the sip phone.

The SIP part is simple:

soft phone - asterisk: INVITE

asterisk - soft phone: TRYING

asterisk - provider: INVITE

asterisk - soft phone: 180 RINGING

provider - asterisk: 183 SESSION PROGRESS

provider - asterisk: AUDIO

Unfortunately the AUDIO received from the provider by the asterisk box is
not sent to the soft phone.

I think I have tried every combination of progressinband and
prematuremedia, without success.

How can I made the audio received from the provider to the asterisk be
transmitted to the soft phone?

Thank you

Leandro

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Re: [asterisk-users] Routing premature media to the calling channel

[Alex Balashov]

Are you absolutely sure that nothing is coming out, even on a different 
interface than the one on which you are capturing?  Are you capture on the 
Asterisk server and not the receiving host?

Secondly, are you absolutely positive that something is supposed to be coming 
out?  183 does not logically imply or mandate backward early media, though 
183+SDP is generally used as a convention to indicate that it is about to be 
sent.  

--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0671
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

Leandro Dardini ldard...@gmail.com wrote:

All NAT and firewall problems are already been excluded. All peers are on
public IP address and no firewall is active between them. The missing
routing of the audio path to the peer has been checked with tcpdump ...
nothing is coming out from the asterisk box.

Leandro

2012/3/25 Alex Balashov abalas...@evaristesys.com

 I assume you have ruled out NAT and firewall issues?

 Between those two, 99% of the reasons why something may not be routed
 somewhere correctly are accounted for.

 If you donapos;t know, your best bet is to take a packet capture or SIP
 debug on the Asterisk server and find out where that early media is going.

 --
 Alex Balashov - Principal
 Evariste Systems LLC
 235 E Ponce de Leon Ave
 Suite 106
 Atlanta, GA 30030
 Tel: +1-678-954-0671
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com


 Leandro Dardini ldard...@gmail.com wrote:

 Hello,
 I have a problem with premature media and inband progress audio. I am
 using the latest 1.8.10.1 and this is the setup:

 soft phone --- asterisk --- SIP provider

 The number I call is giving back some hints via inband audio I am not able
 to ear from the soft phone. They stop on the asterisk and are not routed
 down the path to the sip phone.

 The SIP part is simple:

 soft phone - asterisk: INVITE

 asterisk - soft phone: TRYING

 asterisk - provider: INVITE

 asterisk - soft phone: 180 RINGING

 provider - asterisk: 183 SESSION PROGRESS

 provider - asterisk: AUDIO

 Unfortunately the AUDIO received from the provider by the asterisk box is
 not sent to the soft phone.

 I think I have tried every combination of progressinband and
 prematuremedia, without success.

 How can I made the audio received from the provider to the asterisk be
 transmitted to the soft phone?

 Thank you

 Leandro



 --
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Re: [asterisk-users] Routing premature media to the calling channel

[Alex Balashov]

I think I may have misunderstood your initial question, sorry.

You are looking for Asterisk to directly pass through the early media from 
upstream?  Why would it do that?

--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0671
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

Leandro Dardini ldard...@gmail.com wrote:

The asterisk box has only one interface. I am capturing all the traffic on
the box and the only audio traffic is from the provider to the asterisk box.

Obviously if I set progressinband=yes, then I get the ringing tone from the
asterisk box, but no the audio from the provider I was looking for.

Leandro

2012/3/25 Alex Balashov abalas...@evaristesys.com

 Are you absolutely sure that nothing is coming out, even on a different
 interface than the one on which you are capturing? Are you capture on the
 Asterisk server and not the receiving host?

 Secondly, are you absolutely positive that something is supposed to be
 coming out? 183 does not logically imply or mandate backward early media,
 though 183+SDP is generally used as a convention to indicate that it is
 about to be sent.

 --
 Alex Balashov - Principal
 Evariste Systems LLC
 235 E Ponce de Leon Ave
 Suite 106
 Atlanta, GA 30030
 Tel: +1-678-954-0671
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com

 Leandro Dardini ldard...@gmail.com wrote:

 All NAT and firewall problems are already been excluded. All peers are on
 public IP address and no firewall is active between them. The missing
 routing of the audio path to the peer has been checked with tcpdump ...
 nothing is coming out from the asterisk box.

 Leandro

 2012/3/25 Alex Balashov abalas...@evaristesys.com

 I assume you have ruled out NAT and firewall issues?

 Between those two, 99% of the reasons why something may not be routed
 somewhere correctly are accounted for.

 If you donapos;t know, your best bet is to take a packet capture or SIP
 debug on the Asterisk server and find out where that early media is going.

 --
 Alex Balashov - Principal
 Evariste Systems LLC
 235 E Ponce de Leon Ave
 Suite 106
 Atlanta, GA 30030
 Tel: +1-678-954-0671
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com


 Leandro Dardini ldard...@gmail.com wrote:

 Hello,
 I have a problem with premature media and inband progress audio. I am
 using the latest 1.8.10.1 and this is the setup:

 soft phone --- asterisk --- SIP provider

 The number I call is giving back some hints via inband audio I am not
 able to ear from the soft phone. They stop on the asterisk and are not
 routed down the path to the sip phone.

 The SIP part is simple:

 soft phone - asterisk: INVITE

 asterisk - soft phone: TRYING

 asterisk - provider: INVITE

 asterisk - soft phone: 180 RINGING

 provider - asterisk: 183 SESSION PROGRESS

 provider - asterisk: AUDIO

 Unfortunately the AUDIO received from the provider by the asterisk box is
 not sent to the soft phone.

 I think I have tried every combination of progressinband and
 prematuremedia, without success.

 How can I made the audio received from the provider to the asterisk be
 transmitted to the soft phone?

 Thank you

 Leandro



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



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Re: [asterisk-users] Routing premature media to the calling channel

[Alex Balashov]

As far as I know, this is not the general tendency of any B2BUA that generates 
such media independently.  However, I could be mistaken.

--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0671
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

Leandro Dardini ldard...@gmail.com wrote:

I want to have the early media to pass from the provider down to the soft
phone because it contains important information about the call, like Your
call cannot go through, please try your call again  ... The provider is
giving this info via early media, just after the 183 SESSION PROGRESS.

Leandro

2012/3/25 Alex Balashov abalas...@evaristesys.com

 I think I may have misunderstood your initial question, sorry.

 You are looking for Asterisk to directly pass through the early media from
 upstream? Why would it do that?


 --
 Alex Balashov - Principal
 Evariste Systems LLC
 235 E Ponce de Leon Ave
 Suite 106
 Atlanta, GA 30030
 Tel: +1-678-954-0671
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com

 Leandro Dardini ldard...@gmail.com wrote:

 The asterisk box has only one interface. I am capturing all the traffic on
 the box and the only audio traffic is from the provider to the asterisk box.

 Obviously if I set progressinband=yes, then I get the ringing tone from
 the asterisk box, but no the audio from the provider I was looking for.

 Leandro

 2012/3/25 Alex Balashov abalas...@evaristesys.com

 Are you absolutely sure that nothing is coming out, even on a different
 interface than the one on which you are capturing? Are you capture on the
 Asterisk server and not the receiving host?

 Secondly, are you absolutely positive that something is supposed to be
 coming out? 183 does not logically imply or mandate backward early media,
 though 183+SDP is generally used as a convention to indicate that it is
 about to be sent.

 --
 Alex Balashov - Principal
 Evariste Systems LLC
 235 E Ponce de Leon Ave
 Suite 106
 Atlanta, GA 30030
 Tel: +1-678-954-0671
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com

 Leandro Dardini ldard...@gmail.com wrote:

 All NAT and firewall problems are already been excluded. All peers are on
 public IP address and no firewall is active between them. The missing
 routing of the audio path to the peer has been checked with tcpdump ...
 nothing is coming out from the asterisk box.

 Leandro

 2012/3/25 Alex Balashov abalas...@evaristesys.com

 I assume you have ruled out NAT and firewall issues?

 Between those two, 99% of the reasons why something may not be routed
 somewhere correctly are accounted for.

 If you donapos;t know, your best bet is to take a packet capture or SIP
 debug on the Asterisk server and find out where that early media is going.

 --
 Alex Balashov - Principal
 Evariste Systems LLC
 235 E Ponce de Leon Ave
 Suite 106
 Atlanta, GA 30030
 Tel: +1-678-954-0671
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com


 Leandro Dardini ldard...@gmail.com wrote:

 Hello,
 I have a problem with premature media and inband progress audio. I am
 using the latest 1.8.10.1 and this is the setup:

 soft phone --- asterisk --- SIP provider

 The number I call is giving back some hints via inband audio I am not
 able to ear from the soft phone. They stop on the asterisk and are not
 routed down the path to the sip phone.

 The SIP part is simple:

 soft phone - asterisk: INVITE

 asterisk - soft phone: TRYING

 asterisk - provider: INVITE

 asterisk - soft phone: 180 RINGING

 provider - asterisk: 183 SESSION PROGRESS

 provider - asterisk: AUDIO

 Unfortunately the AUDIO received from the provider by the asterisk box
 is not sent to the soft phone.

 I think I have tried every combination of progressinband and
 prematuremedia, without success.

 How can I made the audio received from the provider to the asterisk be
 transmitted to the soft phone?

 Thank you

 Leandro



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 _
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Re: [asterisk-users] Rate sheet normalization

[Alex Balashov]

Our system just rolls over until it finds a carrier that will take it. Up to 30 
different routes are supported, and rollover is pretty instantaneous in most 
cases.

--
This message was painstakingly thumbed out on my mobile, so apologies for 
brevity and errors.

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

On Mar 15, 2012, at 11:14 PM, Ast Coder asteriskcod...@gmail.com wrote:

 I would be more interested in a system where quality routes are tested with 
 different providers because rate really doesn't matter if a call can't be 
 placed or if a destination is a fake one. We have seen many fake destinations 
 with top tier providers but they had the best rates so the strategy to pick 
 them first really didn't work.
 
 So, maybe a subscription service where a dialler system continuously tests 
 routes with a list of 10 providers so that it's established which routes 
 actually work and then allow that data to be downloaded for usage.
 
 
 
 On Thu, Mar 15, 2012 at 8:42 PM, Markus unive...@truemetal.org wrote:
 Am 15.03.2012 17:20, schrieb Raj Mathur (राज माथुर):
 
 On Thursday 15 Mar 2012, Markus wrote:
 With like 10 different ratesheets from 10 different providers, of
 which many change their rates every few days, manually doing it in
 Excel is too time consuming...
 
 Is it possible to get samples?  I'd be interested in looking into
 developing a script that can handle this problem generically, and
 presumably you're available to alpha- and beta-test in any case :)
 
 Most definitely! I'll get in touch off-list. :)
 
 
 
 
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Re: [asterisk-users] DAHDISendCallreroutingFacility

[Alex Villací­s Lasso]

El 10/03/12 12:05, Karsten Wemheuer escribió:

Hi,

Am Samstag, den 10.03.2012, 08:42 -0800 schrieb Mehdi Shirazi:

Hi
I installed Asterisk 1.8.7 with CD ISO(Elastix 2.2)
I want to use DAHDISendCallreroutingFacility Application on a PRI link(LIBPRI 
Already installed).
according to
https://wiki.asterisk.org/wiki/display/AST/New+in+1.8
Asterisk 1.8 include this application but I cannot see it with core show 
applications
Do I need to install mISDN or other modules for using that ?

Regards
M.Shirazi

No You don't need mISDN or other modules. DAHDISendCallreroutingFacility
is part of chan_dahdi. But as far as I know Asterisk 1.8.7 had problems
with this application. Try using at least 1.8.8 (1.8.10.0 is currently
the stable version of 1.8 release).

HTH,
Karsten



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Elastix has 1.8.10.0 in the elastix-beta repository (yum 
--enablerepo=elastix-beta).

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