externip nat audio sip trunk issue problem

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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…

Asterisk Users 3.5 years ago 3 Answers

Is this doable?

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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)…

Asterisk Users 3.5 years ago 7 Answers

Getting one way audio even NAT is configured

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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
>…

Asterisk Users 3.5 years ago 1 Answer

Dynamically toggling ConfBridge recording from conference menu

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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…

Asterisk Users 3.5 years ago 1 Answer

Asterisk 10.1.2 Now Available

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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…

Asterisk Users 3.5 years ago 0 Answers

Asterisk 1.8.9.2 Now Available

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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…

Asterisk Users 3.5 years ago 1 Answer

SIP Provider Russia, Ukraine, Poland

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Hello List! I'm searching for SIP-Providers in the following countries:
Russia
Ukraine
Poland I need a geographical number for each country, maybe a prepaid
SIP-Account, trunking is not important.
Has anyone some experience with these countries? yours
christian

Asterisk Users 3.5 years ago 2 Answers

Asterisk 10.0 Realtime

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Hi I have noticed new behaviour of asterisk 10.0 realtime.
In 1.6 when I was using realtime: """
[somecontext] exten => someexten1......
exten => someexten2......
exten => someexten3......
exten => someexten4...... switch => Realtime/${CONTEXT}@extensions
""" switch statement was executed after lines above (so there was a
precedence of the lines declared in a extensions.conf over the ones in
database). In asterisk 10.0 switch is executed before extens declared in the
extensions.conf file. Is there a way to change that and have previous behaviour?
Cheers

Asterisk Users 3.5 years ago 1 Answer

read digits during recording / DTMF in conference?

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Hi, I want to create a system for incoming calls where, under some
circumstances, callers get routed straight to voicemail (or some other
means of recording a message) but if they enter a valid extension number
then the recorded message would be abandoned and they'd be diverted to
the extension number they entered. I realise this can be done with the voicemail app with operator=yes but
the problem with this is that the caller has to press 0 while the
announcement is being played. If they're too slow and recording has
started, they've…

Asterisk Users 3.5 years ago 3 Answers