* You are viewing the archive for February 1st, 2012

externip nat audio sip trunk issue problem

On Wed, Feb 1, 2012 at 9:14 PM, Gabriel Ortiz Lour
wrote:
> Hi all,
>
>   I’ve tried search this problem on the list… no luck…
>
>   The case is:
>
> without externip/localnet config on sip.conf [general] my SIP trunk works,
> but with no audio NAT problem (asterisk sends the private 192 address to the
> outside…)
>
> when I configure externip/localnet correctly my SIP trunk simply disappear!
> Checking the signalling with tcpdump shows me that Im sending the packets to
> the correct SIP trunk IP but there is no response AT ALL from it…

Can you explain this?
What do you mean no response? Is it registering? Do you have a debug output?

>
> Anyone had this problem?
>
> Thanks,
> Gabriel
>
> –
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Is this doable?

Whats asterick?

On Wed, Feb 1, 2012 at 7:48 PM, Josh wrote:
> I am trying to configure Asterick, having the following system setup on
> the Asterick server:
>
> * eth0 faces the external Internet interface, *but* it does not have IP
> address (it has a private one given to it by my ISP’s DHCP server);
> * eth1 faces my internal network (say 10.1.1.0/24);
> * tun0 serves all mobile smartphones and connects to the internal
> network (it has a different ip range, say 10.1.2.0/24) – they are all
> connected via the Internet using OpenVPN;
>
> I would like to configure Asterick for internal calls between ourselves
> (eth1< ->tun0) and I think I have no problem with configuring this part.
> I would also like to use one external VOIP provider to which Asterick
> registers on startup. I think I know how to do that and use the
> “register” option in sip.conf, though I am not sure for the rest of the
> NAT-related entries (see below).
>
> The purpose of registering this external account is so that both the
> smart phones (tun0) and the internal net (eth1) users could use this
> account to make external calls (starting with “0”, i.e “_0[0-9].”
> pattern in extensioins.conf). Obviously, I need these calls to be routed
> properly via the external VOIP account. In addition to that, I would
> also need to receive calls from that external account to a nominated
> internal one (say on extension 20).
>
> Is this achievable?
>
> If so, I am not completely clear on whether I need to explicitly specify
> my public IP address (via externip/externhost) or whether Asterick is
> able to find it without this option? If not, then my plan is to use
> external program to find it and then use a script in Asterick to set it
> up as an environment variable. Would that work? That external IP address
> is going to change, but only in rare circumstances and in such cases I
> have to restart a lot of stuff (including Asterick) on that server (this
> is usually triggered by a monitoring program), so it won’t be a problem
> once it is setup initially. I am also not sure whether to specify
> “nat=yes” or just have “nat=route” only – any ideas?
>
> Is there a comprehensive list of all the options available in sip.conf
> and what they do, because I was unable to find such a list?
>
> If the above is doable, I would also like to add the following 2 features:
>
> 1. Secondary external VOIP account, though I have no idea how to specify
> its port in “register” (it uses port 5065 instead of the standard 5060).
> That account would need to be used on a separate interface (eth2) with a
> different public IP address. Would it be possible to use
> externip/externhost inside that external account section to specify it?
> If this is not possible, then I am thinking of running a separate
> instance of Asterick with the second VOIP account/public IP address set
> up – would that work?
>
> 2. I would like to be able to configure the following work flow: for a
> specific set of (external) calling numbers (including where no Caller ID
> is available):
> a) these callers to be prompted to specify the “reason” for their call;
> b) their response to be temporarily “recorded”/stored (a short message
> of, say no more than 10 seconds long or when they press ‘#’ for that
> recording to stop);
> c) Asterick then rings the nominated number for external VOIP calls
> (extension 20) and play that recorded message back;
> d) then asks for one of four possible outcomes:
> – accept this call (pressing, say 1) in which case the call is connected
> as normal;
> – reject it with a message that that number/person is “unavailable”
> (say, by pressing 0);
> – ask the caller to leave a message by transferring them to a voicemail
> (say by pressing 2); or
> – end the initial call completely with a message that the caller/number
> has been “blacklisted” (say, by pressing the 9 key);
>
> Could this be achieved?
>
> One final question about binding: in order to be able to use both tun0
> and eth1 interfaces so that Asterick serves the calls from both eth1 and
> tun0, do I have to use “bind 0.0.0.0″? Is there an alternative, like
> specifying “bind 10.1.1.1″ for eth1 and then “bind 10.1.2.1″ for the
> tun0 interface – is this possible?
>
> Many thanks in advance!
>
>
> –
> _____________________________________________________________________
> — 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
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>  http://lists.digium.com/mailman/listinfo/asterisk-users

Getting one way audio even NAT is configured

On Wed, Feb 1, 2012 at 1:16 PM, Ahmed Munir wrote:

> Hi all,
>
> I’m getting one way audio when calling over the SIP trunk i.e. end device
> B (remote end of SIP trunk) can hear device A (softphone registered with
> Asterisk) but device A can’t hear device B. Even though I configured same
> NAT configurations on other servers and they are working good. The NAT
> configuration is listed below;
>
> localnet=130.0.0.0/130.0.0.0
> externhost=12.131.12.13
> externrefresh=10
> fromdomain=test.localhost.com
> nat=yes
> qualify=yes
> canreinvite=no
>
>
> NAT on device end i.e. my softphone (extension) has already set to yes
> with canreinvite=no but still unable to resolve this issue. SIP traces are
> listed below;
>
>

>
> The Asterisk version I’m using is 1.8.5. Please assist me at earliest.
>

Which device (A or B) is behind NAT with regards to your asterisk server?
Is that the actual localnet= statement you’re using, because to my
understanding that is not the proper format to use (should be
localnet=x.x.x.x/y.y.y.y where x.x.x.x is your actual local network, and
y.y.y.y is your subnet for your local network).

Dynamically toggling ConfBridge recording from conference menu

Hello,

I’m using ConfBridge in an application where I need a conference admin
to be able to start and stop recording using a conference menu option.

Currently, I’m doing this by defining ConfBridge menu options

7=dialplan_exec(conference_functions,rec_start,1)
9=dialplan_exec(conference_functions,rec_stop,1)

The rec_start and rec_stop extensions simply start/stop MixMonitor on
the channel of the admin who presses 7/9. However, what I’d really like
to do is to be able to execute the equivalent of the CLI “confbridge
record start xxxx” command, so that the recording would be independent
of the participant channel.

I suppose I could do this with a System call, something like
System(asterisk -rx “confbridge record start xxxx”) – but is there a
better, less-roundabout way of getting there?

Thanks,
Josh

Asterisk 10.1.2 Now Available

The Asterisk Development Team has announced the release of Asterisk 10.1.2. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 10.1.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

* — Fix SIP INFO DTMF handling for non-numeric codes —
(Closes issue ASTERISK-19290. Reported by: Ira Emus)

* — Fix crash in ParkAndAnnounce —
(Closes issue ASTERISK-19311. Reported-by: tootai)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.2

Thank you for your continued support of Asterisk!

Asterisk 1.8.9.2 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.8.9.2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.9.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

* — Fix SIP INFO DTMF handling for non-numeric codes —
(Closes issue ASTERISK-19290. Reported by: Ira Emus)

* — Fix crash in ParkAndAnnounce —
(Closes issue ASTERISK-19311. Reported-by: tootai)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.2

Thank you for your continued support of Asterisk!