Re: [asterisk-users] Sending calls from behind NAT

[bilal ghayyad]

Dears;

What Jian said is the right and it worked.

But I have the following questions:

Why 192.168.10.2 is wrong and I have to use 192.168.10.0? Also, do I have to 
set the localnet or it is enough to set the externip?

>From the other side, I am using Asterisk 1.8.12.0 and when I was searching in 
>the sip.conf, I did not find externip (so I added by my hand) and I remember 
>very well that before I was able to find the externip in the sip.conf, 
>although I am finding externadd. So why this? 

One more thing, what is the difference between externadd and externip?

Regards
Bilal

---
> > Dears;
> >
> > It seems my service provider is requesting a
> complicated settings to allow me to send from behind NAT.
> >
> > What they said:
> >
> > "It shouldn't matter as long as you are handling the
> NAT correctly your end. We do not fix NAT so if you're
> sending internal addresses in your INVITEs or SDP then
> things will fail but if you're handling it correctly, we
> shouldn't tell the difference".
> >
> >
> > Really, I did not understand what exactly they need.
> But maybe what they need is to see my public IP address
> without the private IP address (this what I understood if I
> am right).
> >
> > I tried to use the following in the [general] settings
> in the sip.conf
> >
> > localnet=192.168.10.2/255.255.255.254
> > externadd =196.40.164.239
> >
> 
> 
> I think these setting are all wrong:
> 1. local network should be something like: 192.168.10.0
> 2. Subnetmask cant' be 255.255.255.254 !
> 3. externip=x.x.x.x (Not "externadd")
> 
> Jian
> 
> > But even, the calls are drop .. so what I have to do?
> >
> > The following what I get when I enabled the sip debug:
> >
> >
> > <--- SIP read from UDP:194.0.220.220:5060 --->
> > SIP/2.0 403 UA behind NAT not accepted here
> > Via: SIP/2.0/UDP
> 192.168.10.2:5060;branch=z9hG4bK123c8781;rport=5060;received=196.40.164.239
> > From: "asterisk"
> ;tag=as45d7c63b
> > To: 
> > ;tag=84f31a80f7bc633204d0bd8d76e9cb24.53a5
> > Call-ID:
> 6f68f414109ba1ff596726c74a0cb5ac@192.168.10.2:5060
> > CSeq: 102 INVITE
> > P-Behind-NAT: source
> > Server: Service Provider Global Proxy v2
> > Content-Length: 0
> >
> > So what could resolve my problem?
> >
> > Regards
> > Bilal


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Astricon 2012 presentations

[Andrew White]

Hey Dan,

Please keep us updated on a video or transcript of this talk - this seems like 
a very fascinating presentation and I'd love to get more information.

Cheers,

Andrew.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Jenkins
Sent: Monday, 12 November 2012 10:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Astricon 2012 presentations

Hi,

As far as I'm aware the videos are still being produced and there's no 
definitive list anywhere for the slide decks.

However, my one is here: 
http://www.slideshare.net/danjenkins/asterisk-html5-and-nodejs-a-world-of-endless-possibilities-14881614

Dan Jenkins

--
Dan Jenkins - Senior Web Developer
email: dan.jenk...@holidayextras.com
twitter: dan_jenkins
linkedin: jenkinsdaniel
skype: d-jenkins
blog: www.dan-jenkins.co.uk
about.me: about.me/dan_jenkins


On 12 November 2012 11:05, Lenz Emilitri 
mailto:lenz.lo...@gmail.com>> wrote:
Hello all,
anybody knows if the PDFs for presentations held at Astricon 2012 are available 
somewhere? I looked at the website but cannot find anything.
Thanks
l.


--
Loway - home of QueueMetrics - http://queuemetrics.com
Test-drive WombatDialer beta @ http://wombatdialer.com


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 1.8.16 Monitoring tools

[Andrew White]

Hey Motty,

The simplest way I've found is having an asterisk console open (asterisk -r) 
with verbosity to level 12. Alternatively you could tail -f the full log (in 
/var/log/asterisk) - I like to parse it with something like ccze to colour code 
things.

The better solution I've found is to use MySQL (or the equivalent database 
program you like) realtime to store my queues and use AGI or recode into LUA to 
log outbound calls to the same database. There is some fairly basic information 
on this at http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL. Once 
you've got all your data sorted how you want, you can develop whatever frondend 
application you want - perl scripts to monitor for trigger events, a nice web 
interface - the possibilities are endless!

Best of luck,

Andrew

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Saturday, 10 November 2012 8:24 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Asterisk 1.8.16 Monitoring tools

Hello,
I want to monitor my Asterisk 1.8, inbound, outbound, status calls, queue call? 
Any suggestions? 

I found Monast, I'm having issues configurating. 

Thanks, 


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to 
Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [asterisk-biz] Service Provider Platform?

[Alistair Cunningham]

Hello Marshall,

Please see Enswitch:

https://integrics.com/enswitch/

It provides everything you ask for except full IAX support (we recommend 
SIP for client connections) and full control of codecs.


It also supports much much more, such as billing, invoicing, payment 
collection, etc, etc. The PBX features and billing are truly integrated, 
and truly multi-tenant at the application level. It runs on highly 
scalable and fully redundant clusters. Failover between sites is 
supported given a suitable network, and a couple of our customers are 
doing this. Many of our customers have come from an environment where 
they're providing hosted PBX services with multiple instances of a PBX 
product such as FreePBX, and who want to expand to something that 
supports tens to tens of thousands of customers with hundreds to 
hundreds of thousands of users. The largest system has over 150,000 
users. We also have some customers using Enswitch as a corporate PBX for 
major institutions (such as financial services companies) who have users 
distributed across many sites.


If interested, please drop me an email off-list and we can discuss 
system sizing, pricing, support, etc.


On 13/11/12 22:37, Marshall Henderson wrote:

Hey guys and gals-

Right now, I'm using FreePBX to handle providing voice services to a
handful of customers. However, it just isn't cutting it for features,
billing, customer access (portal stuff), etc. What do you recommend? Is
there an ITSP portal/panel/platform available for running an ITSP with
Asterisk?

Features I'm interested in:

-Ability to sell traditional hosted PBX services (handsets register) as
well as 'trunking' (PBX registers)
-Portal access for customers for things like call forwarding, routing
management, billing access, CDRs, maybe even configuration of service
such as codecs/protocol
-LCR for multiple providers/ratedecks/routes
-Full support for both SIP and IAX2

Also, looking for something with built in failover as I'd like to have a
replicated system in a second colo facility to handle calls if one colo
is down.

Thoughts? Suggestions? Product examples? Am I dreaming and nothing like
the above exists?

Sorry for the x-post between lists, hoping to get the largest number of
responses from all who frequent here.

mch



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz



--
Alistair Cunningham
+1 888 468 3111
+44 20 799 39 799
http://integrics.com/

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Service Provider Platform?

[Marshall Henderson]

Hey guys and gals-

Right now, I'm using FreePBX to handle providing voice services to a
handful of customers. However, it just isn't cutting it for features,
billing, customer access (portal stuff), etc. What do you recommend? Is
there an ITSP portal/panel/platform available for running an ITSP with
Asterisk?

Features I'm interested in:

-Ability to sell traditional hosted PBX services (handsets register) as
well as 'trunking' (PBX registers)
-Portal access for customers for things like call forwarding, routing
management, billing access, CDRs, maybe even configuration of service such
as codecs/protocol
-LCR for multiple providers/ratedecks/routes
-Full support for both SIP and IAX2

Also, looking for something with built in failover as I'd like to have a
replicated system in a second colo facility to handle calls if one colo is
down.

Thoughts? Suggestions? Product examples? Am I dreaming and nothing like the
above exists?

Sorry for the x-post between lists, hoping to get the largest number of
responses from all who frequent here.

mch
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] On SIP REGISTER event trigger a AGI script

[Steve Edwards]

On Wed, 14 Nov 2012, Face wrote:


Is there a way I can  trigger a AGI script On SIP REGISTER event.



On Wed, Nov 14, 2012 at 2:38 AM, Steve Edwards


Well, an AGI runs in the context of a channel. A REGISTER does not.

So, no.


On Wed, 14 Nov 2012, Face wrote:


Is there a way to accomplish my goal ?



On Wed, Nov 14, 2012 at 2:38 AM, Steve Edwards


I don't know what your goal is.

An AGI cannot execute without a channel and a REGISTER does not create a 
channel.


Are you are asking 'Can I execute something when Asterisk receives a 
REGISTER request?'


I'm just a 1.2 Luddite, but I suspect listening to events on the manager 
interface (AMI) from an external daemon would be the way to go.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] On SIP REGISTER event trigger a AGI script

[Face]

On Wed, Nov 14, 2012 at 2:38 AM, Steve Edwards
 wrote:
> On Wed, 14 Nov 2012, Face wrote:
>
>> Is there a way I can  trigger a AGI script On SIP REGISTER event.
>
>
> Well, an AGI runs in the context of a channel. A REGISTER does not.
>
> So, no.
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

Is there a way to accomplish my goal ?

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] On SIP REGISTER event trigger a AGI script

[Steve Edwards]

On Wed, 14 Nov 2012, Face wrote:


Is there a way I can  trigger a AGI script On SIP REGISTER event.


Well, an AGI runs in the context of a channel. A REGISTER does not.

So, no.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] On SIP REGISTER event trigger a AGI script

[Face]

Hello All,

Is there a way I can  trigger a AGI script On SIP REGISTER event.

-- 
Any help would be much appreciated.
falazemi

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Noise on phones while speaking...

[Carlos Chavez]

On 11/13/12 4:31 PM, Mark Engelhardt wrote:

Carlos,

I think the noise you are hearing might echo cancelation that is broken or set 
incorrectly. Maybe the card and asterisk are both trying to echo cancel?

Mark

On Nov 13, 2012, at 1:52 PM, Carlos Chavez wrote:


I have a new install and the customer is complaining that they hear noise 
on all calls, no matter if it is internal or external, desk phones or 
softphones.  The noise is only present when the user is speaking, not the 
remote side.  The remote side does not hear the noise, only the local user.

We are using Asterisk .1.8.11-cert8 on a CentOS 6 machine with a Digium 
AEX800 card and DAHDI 2.6.1.  I really do not know how this noise is generated. 
 Where can I look?  Why would a SIP to SIP call have this noise?


The card itself does not have hardware echo cancellation so we use 
MG2.  I am not fixated on the card because this should not affect a SIP 
to SIP internal call unless the card is really defective and provides 
bad timing to Asterisk.


--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Noise on phones while speaking...

[Mark Engelhardt]

Carlos,

I think the noise you are hearing might echo cancelation that is broken or set 
incorrectly. Maybe the card and asterisk are both trying to echo cancel?

Mark

On Nov 13, 2012, at 1:52 PM, Carlos Chavez wrote:

>I have a new install and the customer is complaining that they hear noise 
> on all calls, no matter if it is internal or external, desk phones or 
> softphones.  The noise is only present when the user is speaking, not the 
> remote side.  The remote side does not hear the noise, only the local user.
> 
>We are using Asterisk .1.8.11-cert8 on a CentOS 6 machine with a Digium 
> AEX800 card and DAHDI 2.6.1.  I really do not know how this noise is 
> generated.  Where can I look?  Why would a SIP to SIP call have this noise?
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>  http://www.asterisk.org/hello
> 
> 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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Sending calls from behind NAT

[J Gao]

On 12-11-13 01:16 PM, bilal ghayyad wrote:

Dears;

It seems my service provider is requesting a complicated settings to allow me 
to send from behind NAT.

What they said:

"It shouldn't matter as long as you are handling the NAT correctly your end. We do 
not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things 
will fail but if you're handling it correctly, we shouldn't tell the difference".


Really, I did not understand what exactly they need. But maybe what they need 
is to see my public IP address without the private IP address (this what I 
understood if I am right).

I tried to use the following in the [general] settings in the sip.conf

localnet=192.168.10.2/255.255.255.254
externadd =196.40.164.239




I think these setting are all wrong:
1. local network should be something like: 192.168.10.0
2. Subnetmask cant' be 255.255.255.254 !
3. externip=x.x.x.x (Not "externadd")

Jian


But even, the calls are drop .. so what I have to do?

The following what I get when I enabled the sip debug:


<--- SIP read from UDP:194.0.220.220:5060 --->
SIP/2.0 403 UA behind NAT not accepted here
Via: SIP/2.0/UDP 
192.168.10.2:5060;branch=z9hG4bK123c8781;rport=5060;received=196.40.164.239
From: "asterisk" ;tag=as45d7c63b
To: 
;tag=84f31a80f7bc633204d0bd8d76e9cb24.53a5
Call-ID: 6f68f414109ba1ff596726c74a0cb5ac@192.168.10.2:5060
CSeq: 102 INVITE
P-Behind-NAT: source
Server: Service Provider Global Proxy v2
Content-Length: 0

So what could resolve my problem?

Regards
Bilal

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Sending calls from behind NAT

[Pat Collins]

Check reinvite and NAT settings on the line as well as the SIP peers.

You can use a stun client from inside your network to see what’s going on with 
the NAT

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Budinick
Sent: Tuesday, November 13, 2012 4:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sending calls from behind NAT

 

I'm with Duncan, you need a public IP address, not private. 

Chris Budinick

Network Technician

RAINIER CONNECT





  _  

From: "Duncan Turnbull" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Tuesday, November 13, 2012 1:29:28 PM
Subject: Re: [asterisk-users] Sending calls from behind NAT


On 14/11/2012, at 10:16 AM, bilal ghayyad  wrote:

> Dears;
> 
> It seems my service provider is requesting a complicated settings to allow me 
> to send from behind NAT. 
> 
> What they said:
> 
> "It shouldn't matter as long as you are handling the NAT correctly your end. 
> We do not fix NAT so if you're sending internal addresses in your INVITEs or 
> SDP then things will fail but if you're handling it correctly, we shouldn't 
> tell the difference".
> 
> 
> Really, I did not understand what exactly they need. But maybe what they need 
> is to see my public IP address without the private IP address (this what I 
> understood if I am right).
> 
> I tried to use the following in the [general] settings in the sip.conf
> 
> localnet=192.168.10.2/255.255.255.254
> externadd =196.40.164.239
> 
This should be externip not externadd

You are still sending them your local address

Cheers Duncan


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Sending calls from behind NAT

[Chris Budinick]

I'm with Duncan, you need a public IP address, not private. Chris BudinickNetwork Technician

RAINIER CONNECTFrom: "Duncan Turnbull" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, November 13, 2012 1:29:28 PMSubject: Re: [asterisk-users] Sending calls from behind NATOn 14/11/2012, at 10:16 AM, bilal ghayyad  wrote:> Dears;> > It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. > > What they said:> > "It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the difference".> > > Really, I did not understand what exactly they need. But maybe what they need is to see my public IP address without the private IP address (this what I understood if I am right).> > I tried to use the following in the [general] settings in the sip.conf> > localnet=192.168.10.2/255.255.255.254> externadd =196.40.164.239> This should be externip not externaddYou are still sending them your local addressCheers Duncan--_-- Bandwidth and Colocation Provided by http://www.api-digital.com --New to Asterisk? Join us for a live introductory webinar every Thurs:               http://www.asterisk.org/helloasterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Sending calls from behind NAT

[Chris Bagnall]

On 13/11/12 9:31 pm, Leighton Brennan wrote:

It looks like you need to enable the sip application layer gateway or ALG on 
your router


Quite often the reverse is true. Most routers (at least those I've used) 
seem to have such a lousy implementation of a SIP ALG it's often far 
better to just disable it and do your own NAT fixups in Asterisk (as 
others have indicated in previous posts).


In fact, it's now the first thing we advise clients to do when they 
report call problems or one-way audio: disable the SIP ALG in your router.


Sadly, there are also quite a few routers out there now that have ALGs 
that can't be disabled (or that make it extremely difficult to disable 
them).


Kind regards,

Chris
--
This email is made from 100% recycled electrons

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Sending calls from behind NAT

[Leighton Brennan]

It looks like you need to enable the sip application layer gateway or ALG on 
your router. The problem is not exactly a Nat issue. The problem is most likely 
with the sip header keeping the private IP address, the ALG when enabled will 
change this to your public 

On 13 Nov 2012, at 21:17, "bilal ghayyad"  wrote:

> Dears;
> 
> It seems my service provider is requesting a complicated settings to allow me 
> to send from behind NAT. 
> 
> What they said:
> 
> "It shouldn't matter as long as you are handling the NAT correctly your end. 
> We do not fix NAT so if you're sending internal addresses in your INVITEs or 
> SDP then things will fail but if you're handling it correctly, we shouldn't 
> tell the difference".
> 
> 
> Really, I did not understand what exactly they need. But maybe what they need 
> is to see my public IP address without the private IP address (this what I 
> understood if I am right).
> 
> I tried to use the following in the [general] settings in the sip.conf
> 
> localnet=192.168.10.2/255.255.255.254
> externadd =196.40.164.239
> 
> But even, the calls are drop .. so what I have to do?
> 
> The following what I get when I enabled the sip debug:
> 
> 
> <--- SIP read from UDP:194.0.220.220:5060 --->
> SIP/2.0 403 UA behind NAT not accepted here
> Via: SIP/2.0/UDP 
> 192.168.10.2:5060;branch=z9hG4bK123c8781;rport=5060;received=196.40.164.239
> From: "asterisk" ;tag=as45d7c63b
> To: 
> ;tag=84f31a80f7bc633204d0bd8d76e9cb24.53a5
> Call-ID: 6f68f414109ba1ff596726c74a0cb5ac@192.168.10.2:5060
> CSeq: 102 INVITE
> P-Behind-NAT: source
> Server: Service Provider Global Proxy v2
> Content-Length: 0
> 
> So what could resolve my problem?
> 
> Regards
> Bilal
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> 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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Sending calls from behind NAT

[Duncan Turnbull]

On 14/11/2012, at 10:16 AM, bilal ghayyad  wrote:

> Dears;
> 
> It seems my service provider is requesting a complicated settings to allow me 
> to send from behind NAT. 
> 
> What they said:
> 
> "It shouldn't matter as long as you are handling the NAT correctly your end. 
> We do not fix NAT so if you're sending internal addresses in your INVITEs or 
> SDP then things will fail but if you're handling it correctly, we shouldn't 
> tell the difference".
> 
> 
> Really, I did not understand what exactly they need. But maybe what they need 
> is to see my public IP address without the private IP address (this what I 
> understood if I am right).
> 
> I tried to use the following in the [general] settings in the sip.conf
> 
> localnet=192.168.10.2/255.255.255.254
> externadd =196.40.164.239
> 
This should be externip not externadd

You are still sending them your local address

Cheers Duncan


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Sending calls from behind NAT

[Eliezer Croitoru]

Dear Bilal,

I understood correctly that the problem is that calls drops?
What router are you using?

Eliezer

On 11/13/2012 11:16 PM, bilal ghayyad wrote:

Dears;

It seems my service provider is requesting a complicated settings to allow me 
to send from behind NAT.

What they said:

"It shouldn't matter as long as you are handling the NAT correctly your end. We do 
not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things 
will fail but if you're handling it correctly, we shouldn't tell the difference".


Really, I did not understand what exactly they need. But maybe what they need 
is to see my public IP address without the private IP address (this what I 
understood if I am right).

I tried to use the following in the [general] settings in the sip.conf

localnet=192.168.10.2/255.255.255.254
externadd =196.40.164.239

But even, the calls are drop .. so what I have to do?

The following what I get when I enabled the sip debug:


<--- SIP read from UDP:194.0.220.220:5060 --->
SIP/2.0 403 UA behind NAT not accepted here
Via: SIP/2.0/UDP 
192.168.10.2:5060;branch=z9hG4bK123c8781;rport=5060;received=196.40.164.239
From: "asterisk";tag=as45d7c63b
To:;tag=84f31a80f7bc633204d0bd8d76e9cb24.53a5
Call-ID:6f68f414109ba1ff596726c74a0cb5ac@192.168.10.2:5060
CSeq: 102 INVITE
P-Behind-NAT: source
Server: Service Provider Global Proxy v2
Content-Length: 0

So what could resolve my problem?

Regards
Bilal



--
Eliezer Croitoru
https://www1.ngtech.co.il
IT consulting for Nonprofit organizations
eliezer  ngtech.co.il

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Sending calls from behind NAT

[bilal ghayyad]

Dears;

It seems my service provider is requesting a complicated settings to allow me 
to send from behind NAT. 

What they said:

"It shouldn't matter as long as you are handling the NAT correctly your end. We 
do not fix NAT so if you're sending internal addresses in your INVITEs or SDP 
then things will fail but if you're handling it correctly, we shouldn't tell 
the difference".


Really, I did not understand what exactly they need. But maybe what they need 
is to see my public IP address without the private IP address (this what I 
understood if I am right).

I tried to use the following in the [general] settings in the sip.conf

localnet=192.168.10.2/255.255.255.254
externadd =196.40.164.239

But even, the calls are drop .. so what I have to do?

The following what I get when I enabled the sip debug:


<--- SIP read from UDP:194.0.220.220:5060 --->
SIP/2.0 403 UA behind NAT not accepted here
Via: SIP/2.0/UDP 
192.168.10.2:5060;branch=z9hG4bK123c8781;rport=5060;received=196.40.164.239
From: "asterisk" ;tag=as45d7c63b
To: 
;tag=84f31a80f7bc633204d0bd8d76e9cb24.53a5
Call-ID: 6f68f414109ba1ff596726c74a0cb5ac@192.168.10.2:5060
CSeq: 102 INVITE
P-Behind-NAT: source
Server: Service Provider Global Proxy v2
Content-Length: 0

So what could resolve my problem?

Regards
Bilal

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Noise on phones while speaking...

[Benny Amorsen]

Carlos Chavez  writes:

> I have a new install and the customer is complaining that they
> hear noise on all calls, no matter if it is internal or external, desk
> phones or softphones.  The noise is only present when the user is
> speaking, not the remote side.  The remote side does not hear the
> noise, only the local user.

If you record the call (with Monitor or Wirehark), does the noise show
up on the recording?


/Benny


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Restarting MOH

[Markus]

Hi Markus,

Am 13.11.2012 20:09, schrieb Markus Weiler:

try to catch in in a cron job per minute.
asterisk -rx 'module unload res_musiconhold.so'


there are always users connected to MOH, except at night, so I still 
would not have a possibility to restart MOH when needed.


Looks like I'm out of luck ...

Thanks!
Markus, too



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Inexpensive SIP Polycom conference phone?

[Ken D'Ambrosio]

Hey, all.  It seems that Polycom has a bunch of offerings for 
conference phones, and I'm just wondering which are the less-expensive 
alternatives; what with their marketing, etc., it's not always obvious 
which is which.


Thanks,

-Ken

P.S.  If anyone's had really good experience with another vendor's 
(relatively inexpensive) conference phone, I'd also be interested in 
hearing about that.


--
This mail was scanned by BitDefender
For more information please visit http://www.bitdefender.com/links/en/frams.html



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Restarting MOH

[Markus Weiler]

hi,

try to catch in in a cron job per minute.

asterisk -rx 'module unload res_musiconhold.so'

Markus




Am 13.11.2012 19:15, schrieb Markus:

Am 13.11.2012 19:01, schrieb Eric Wieling:

module unload res_musiconhold.so
and
module load res_musiconhold.so


Great, that works, but only if no caller is listening to MOH at that 
time. Since *all* my callers are listening to MOH and nothing else, 
that means for me it's the same like an Asterisk restart.


When I try to unload the module I get:

"loader.c:542 ast_unload_resource: Soft unload failed, 
'res_musiconhold.so' has use count 2"


Is there a way to force the unloading?

Any other suggestions?

Thank you!
Markus


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Noise on phones while speaking...

[Chris Bagnall]

On 13/11/12 6:52 pm, Carlos Chavez wrote:

Why would a SIP to SIP call have this noise?


Check to see what random stuff they have on their desk.

We've regularly seen things like mobile phones (or cellphones to those 
of you across the pond :-) ) causing interference with VoIP phones. 
We've also of late seen some (especially Iiyama) monitors doing likewise 
- I suspect they have a fairly noisy 240v-12v transformer inside.


Kind regards,

Chris
--
This email is made from 100% recycled electrons

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Noise on phones while speaking...

[Carlos Chavez]

I have a new install and the customer is complaining that they hear 
noise on all calls, no matter if it is internal or external, desk phones 
or softphones.  The noise is only present when the user is speaking, not 
the remote side.  The remote side does not hear the noise, only the 
local user.


We are using Asterisk .1.8.11-cert8 on a CentOS 6 machine with a 
Digium AEX800 card and DAHDI 2.6.1.  I really do not know how this noise 
is generated.  Where can I look?  Why would a SIP to SIP call have this 
noise?


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk crashing when trying to start a Jabber session with ejabberd

[Patrick Lists]

On 11/13/2012 07:05 PM, Michael L. Young wrote:
[snip]

Is it an omission that this fix has not been applied to the 11 tree?
  From the looks of ASTERISK-19532 it seems that the fix has only been
applied to 1.8 and 10.



If you click on the link for ASTERISK-19532, there is a tab in the Activity section 
labeled "Subversion".  It shows that the patch was applied to 1.8, 10, 11 and 
trunk.


Thanks Michael. Missed that one. Good to know.

Regards,
Patrick


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Restarting MOH

[Markus]

Am 13.11.2012 19:01, schrieb Eric Wieling:

module unload res_musiconhold.so
and
module load res_musiconhold.so


Great, that works, but only if no caller is listening to MOH at that 
time. Since *all* my callers are listening to MOH and nothing else, that 
means for me it's the same like an Asterisk restart.


When I try to unload the module I get:

"loader.c:542 ast_unload_resource: Soft unload failed, 
'res_musiconhold.so' has use count 2"


Is there a way to force the unloading?

Any other suggestions?

Thank you!
Markus


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk crashing when trying to start a Jabber session with ejabberd

[Michael L. Young]

- Original Message -
> From: "Patrick Lists" 
> To: asterisk-users@lists.digium.com
> Sent: Tuesday, November 13, 2012 4:35:54 AM
> Subject: Re: [asterisk-users] Asterisk crashing when trying to start a Jabber 
> session with ejabberd
> 
> On 11/13/2012 12:11 AM, Phil Reynolds wrote:
> [snip]
> > It turns out to be a known issue:
> >
> > https://issues.asterisk.org/jira/browse/ASTERISK-19532
> >
> > ... and can be fixed by applying the patch at:
> >
> > https://issues.asterisk.org/jira/secure/attachment/43441/xmpp_no_crash_with_ejabberd.patch
> >
> > I will file the details with Debian too...
> 
> Is it an omission that this fix has not been applied to the 11 tree?
>  From the looks of ASTERISK-19532 it seems that the fix has only been
> applied to 1.8 and 10.
> 

If you click on the link for ASTERISK-19532, there is a tab in the Activity 
section labeled "Subversion".  It shows that the patch was applied to 1.8, 10, 
11 and trunk.

Michael
(elguero)

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Restarting MOH

[Eric Wieling]

module unload res_musiconhold.so
and
module load res_musiconhold.so

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus
Sent: Tuesday, November 13, 2012 1:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Restarting MOH

Am 13.11.2012 16:51, schrieb OCEANET - Cédric BASSAGET:
> have you tried "module reload res_musiconhold.so" ?

Hi Cedric,

thanks for the suggestion. Unfortunately, it does nothing, just like "moh 
reload".

Any other suggestions?

Regards
Markus



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to 
Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Restarting MOH

[Markus]

Am 13.11.2012 16:51, schrieb OCEANET - Cédric BASSAGET:

have you tried "module reload res_musiconhold.so" ?


Hi Cedric,

thanks for the suggestion. Unfortunately, it does nothing, just like 
"moh reload".


Any other suggestions?

Regards
Markus



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] dahdi firmware for centos 6

[Justin Killen]

In http://packages.digium.com/centos/ there is not yet a centos 6 branch (Nor 
is there a RHEL 6 branch).  Centos 6.0 was release in July of 2011 - is this 
something that Digium is planning on supporting?  Or is there a different URL 
that I'm not aware of for firmware packages?

-Justin Killen
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Astricon 2012 presentations

[Rusty Newton]

On 11/12/2012 5:05 AM, Lenz Emilitri wrote:

Hello all,
anybody knows if the PDFs for presentations held at Astricon 2012 are 
available somewhere? I looked at the website but cannot find anything.

Thanks
l.


I'll try to find out today or tomorrow and post the answer on the list.

--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out at: www.digium.com www.asterisk.org


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Restarting MOH

[OCEANET - Cédric BASSAGET]

have you tried "module reload res_musiconhold.so" ?

Cédric

Le 13/11/2012 16:46, Markus a écrit :

Hi list,

I'm streaming live mp3 streams (web radios) via mplayer and mpg123 
using MusicOnHold. Often, one or two of the streams die during the day 
and I have to restart (not reload) Asterisk to bring it back. "moh 
reload" on the console doesn't do anything.


This is suboptimal if there are active callers, of course.

Is there any way to restart the MOH system without restarting Asterisk?

Asterisk 10.7.1 and 10.8.0.

Thanks!
Markus



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



--
OCEANET
---
[AGENCE DU MANS]
7, rue des Frênes
ZAC de la Pointe
72190 SARGE LES LE MANS
[t] +33 (0)2.43.50.26.50
[f] +33 (0)2.43.72.21.14

[AGENCE D'ANGERS]
5, rue Fleming
Angers Technopole
49066 ANGERS
[t] +33 (0)2.41.19.28.65
[f] +33 (0)2.52.19.22.00

http://www.oceanet.com
http://www.oceanet-telecom.com



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Restarting MOH

[Markus]

Hi list,

I'm streaming live mp3 streams (web radios) via mplayer and mpg123 using 
MusicOnHold. Often, one or two of the streams die during the day and I 
have to restart (not reload) Asterisk to bring it back. "moh reload" on 
the console doesn't do anything.


This is suboptimal if there are active callers, of course.

Is there any way to restart the MOH system without restarting Asterisk?

Asterisk 10.7.1 and 10.8.0.

Thanks!
Markus



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Astricon 2012 presentations

[Ali Pey]

I have also uploaded my presentation here:

http://www.slideshare.net/alipey/astricon-2012-redundancy-and-high-availability?from=share_email

It's on Redundancy and high availability using OpenSIPS/Kamailio.

Regards,
Ali Pey



On Tue, Nov 13, 2012 at 4:24 AM, Lenz Emilitri  wrote:

> Thanks - too bad I missed it :)
>
>
>
> 2012/11/12 Dan Jenkins 
>
>> Hi,
>>
>> As far as I'm aware the videos are still being produced and there's no
>> definitive list anywhere for the slide decks.
>>
>> However, my one is here:
>> http://www.slideshare.net/danjenkins/asterisk-html5-and-nodejs-a-world-of-endless-possibilities-14881614
>>
>> Dan Jenkins
>>
>> --
>> Dan Jenkins - Senior Web Developer
>> email: dan.jenk...@holidayextras.com
>> twitter: dan_jenkins 
>> linkedin: jenkinsdaniel 
>> skype: d-jenkins
>> blog: www.dan-jenkins.co.uk
>> about.me: about.me/dan_jenkins
>>
>>
>>
>> On 12 November 2012 11:05, Lenz Emilitri  wrote:
>>
>>> Hello all,
>>> anybody knows if the PDFs for presentations held at Astricon 2012 are
>>> available somewhere? I looked at the website but cannot find anything.
>>> Thanks
>>> l.
>>>
>>>
>>> --
>>> Loway - home of QueueMetrics - http://queuemetrics.com
>>> Test-drive WombatDialer beta @ http://wombatdialer.com
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>http://www.asterisk.org/hello
>>>
>>> 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 --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Loway - home of QueueMetrics - http://queuemetrics.com
> Test-drive WombatDialer beta @ http://wombatdialer.com
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> 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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] salesforce opencti

[Marek Cervenka]

hello,

do you have someone connector to salesforce?
http://wiki.developerforce.com/page/Open_CTI

i need it for Mac OS X (the web/javascript way, not the old CTI Adapter way)

i'm using Asterisk 1.8

thanks

--
---
Marek Cervenka
===



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to get louder voice ?

[Jacob . E . Miles]

Not sure if this is what you want but you can always set the TX and RX
Gain values via the dialplan.

 

Jacob

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Tuesday, November 13, 2012 4:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to get louder voice ?

 

Hello,

I have the following case.
A customer is a heavy Meetme/audio conference user.
He is equiped with Polycom SS2W (DECT SoundStation 2W audio conference
station).
Users complain they "often do not hear the other party loud enough".

The setup is then:
Remote party <--- PSTN/ISDN---> Asterisk <---SIP---> Kirk300
<---DECT---> SS2W

My questions are:
1. How can I measure audio strength/loudness/quality and strip
social/psychological interferences off ?

2. Is there any builtin mechanism inside Asterisk (this setup is 1.6.1
but upgrade is possible) that can change call volume ?

3. Given my setup is purely digital, could it be the source of calls not
being loud enough ?

4. Suggestions ? Comments

Regards

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] How to get louder voice ?

[Olivier]

Hello,

I have the following case.
A customer is a heavy Meetme/audio conference user.
He is equiped with Polycom SS2W (DECT SoundStation 2W audio conference
station).
Users complain they "often do not hear the other party loud enough".

The setup is then:
Remote party <--- PSTN/ISDN---> Asterisk <---SIP---> Kirk300 <---DECT--->
SS2W

My questions are:
1. How can I measure audio strength/loudness/quality and strip
social/psychological interferences off ?

2. Is there any builtin mechanism inside Asterisk (this setup is 1.6.1 but
upgrade is possible) that can change call volume ?

3. Given my setup is purely digital, could it be the source of calls not
being loud enough ?

4. Suggestions ? Comments

Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk crashing when trying to start a Jabber session with ejabberd

[Patrick Lists]

On 11/13/2012 12:11 AM, Phil Reynolds wrote:
[snip]

It turns out to be a known issue:

https://issues.asterisk.org/jira/browse/ASTERISK-19532

... and can be fixed by applying the patch at:

https://issues.asterisk.org/jira/secure/attachment/43441/xmpp_no_crash_with_ejabberd.patch

I will file the details with Debian too...


Is it an omission that this fix has not been applied to the 11 tree? 
From the looks of ASTERISK-19532 it seems that the fix has only been 
applied to 1.8 and 10.


Regards,
Patrick

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Astricon 2012 presentations

[Lenz Emilitri]

Thanks - too bad I missed it :)


2012/11/12 Dan Jenkins 

> Hi,
>
> As far as I'm aware the videos are still being produced and there's no
> definitive list anywhere for the slide decks.
>
> However, my one is here:
> http://www.slideshare.net/danjenkins/asterisk-html5-and-nodejs-a-world-of-endless-possibilities-14881614
>
> Dan Jenkins
>
> --
> Dan Jenkins - Senior Web Developer
> email: dan.jenk...@holidayextras.com
> twitter: dan_jenkins 
> linkedin: jenkinsdaniel 
> skype: d-jenkins
> blog: www.dan-jenkins.co.uk
> about.me: about.me/dan_jenkins
>
>
>
> On 12 November 2012 11:05, Lenz Emilitri  wrote:
>
>> Hello all,
>> anybody knows if the PDFs for presentations held at Astricon 2012 are
>> available somewhere? I looked at the website but cannot find anything.
>> Thanks
>> l.
>>
>>
>> --
>> Loway - home of QueueMetrics - http://queuemetrics.com
>> Test-drive WombatDialer beta @ http://wombatdialer.com
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> 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 --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Loway - home of QueueMetrics - http://queuemetrics.com
Test-drive WombatDialer beta @ http://wombatdialer.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users