* You are viewing Posts Tagged ‘functionality’

No Audio While Call Forwarding, Yes Audio With Followme

Hi all,

the subject says it all. Technical details:
- Asterisk 1.8.7.1
- Behind NAT
- Using external SIP Provider

The call forwarding is tested both with this functionality on the phone and with configuration in the dialplan. In the latter case a database variable is set to the external number, if set a Dial command calls this number. So really nothing fancy (actually I followed the example on http://www.voip-info.org/wiki/view/Asterisk+call+forwarding ).

sip.conf has nat=yes, externip= … and I tried every setting of directmedia in the providers configuration part.

Followme works flawlessly, so I’m really wondering if this is a NAT issue.


Can anyone point me into a certain direction?


Thx!!!!


BC

Asterisk 11 Queue Calls – Emulate Dial(b) Functionality

I currently run an Asterisk 10 system with hotdesking functionality set up. Several of the users have worked with a system in the past that supported BLF on their IP phones, and would like their current phones to behave in a similar fashion. Right now I have a really kludgy system that mostly works, but doesn’t consistently trigger the cleanup macro to “clear” the device state on the end of a call. Rather than continue to beat my head against the wall playing “which context isn’t firing an h extension to dump calls into the cleanup macro”, I decided to investigate Asterisk 11 for the new Dial() b function and the new hangup handler CHANNEL variable.

I have the hints working more or less correctly on direct calls to/from the phones, making use of the b and U functions in Dial() and some judicious use of GROUP channel variables and CHANNEL(hangup_handler_wipe). But, on my live system, sometimes the users receive calls from a queue, and I don’t see any way with the queue calls to emulate the b functionality in Dial() to be able to set the agent extension’s device state to RINGING when the queue call gets created. Obviously, I can use membergosub to set the agent to “INUSE” after they pick up the call (like Dial() U), but is there anything that I can use to manipulate the channel that is calling the agent while/before it is ringing?

Thank you,

Noah Engelberth MetaLINK Technologies

Please dont tell me this is impossible

I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI->command. I need to play a
file and wait until the customer presses at least $maxdigits to
return, BUT, the file must continue playing until $maxdigits is
received or $timeout has expired. So far I found impossible to achieve
this functionality. Am I missing something?
Philip

Digium IP Phones – Teleworker Capability?

We couldn’t see anything about this on the Digium site, but maybe
someone here can comment?

Do the new Digium phones provide good “teleworker” functionality?

The benchmark we’re comparing against is the capabilities of Mitel
3300 IP systems with Mitel 5330 IP phones (running their proprietary
MINET protocol), specifically:

a. A Mitel phone can be easily configured for teleworker mode (select
TW mode and the IP of the gateway server). The phone reboots and it
is ready to be used (once the Mitel border gateway is set to recognize
the unit’s ID, based on its MAC address, printed on the label on the
back of the phone). If the phone gets reallocated back to a directly
connected office environment, a simple reset procedure brings it back.

b. You can plug in the phone virtually anywhere. It has a built-in
tunnelling mechanism providing end-to-end encryption and is very
tolerant of the network configuration, routers, NAT, etc.

c. If the link between the phone and the gateway goes down, the phone
will restore itself gracefully and automatically once the network
function resumes. Absolutely hassle-free to the user.

d. Users can be configured to have hot-desk functionality. The phone
has a default extension assigned, but the user can be set up so that
they can “log in” to their normal office extension number from
wherever they are. Their office phone is automatically logged-out and
goes to its default extension when you log in to a teleworker phone
(you don’t have to log out from it first). Your phone buttons,
display settings, voicemail WMI and access, (everything) move to this
new phone, and you can work from your home office, on the road, etc.,
and inbound and outbound calls work just like you were there in the
office (callerid, etc).

These four features would be a big selling point for us to consider
moving our organization from Mitel to Digium/Asterisk/Switchvox.

How much of this can be done with Asterisk/Switchvox and, say, the
Digium D70 phone with dynamic button display?

Thanks for all comments!

Adhearsion 2.0 Release For Asterisk 1.8+

Today marks another milestone in the Adhearsion project: the release of Adhearsion 2.0.  There has been a fury of activity in the last few days as we have worked hard to update documentation and release a brand new look-and-feel for the Adhearsion website.  We hope you like it.

So, with a small flourish and no small amount of relief, I’m pleased to announce the immediate availability of Adhearsion 2.0, the open source framework for the creation of voice applications.
Here are some highlights of the changes relative to the latest Adhearsion 1.x:
  • Adhearsion now supports multiple telephony engines! In particular we support Asterisk (as always) as well newly added support for PRISM via the open-standard Rayo protocol
  • CallControllers make telephone functionality more Ruby-esque, more testable and are scientifically shown to make you happier
  • A self-documenting configuration engine (“rake config:show”)
  • A completely revamped plugin system makes adding and sharing Adhearsion functionality better than ever
  • Did I mention the new website design and documentation?
  • Way more stuff than I can reasonably list here.  You should check out the CHANGELOG and the Upgrade documentation.
I would like to take a moment and recognize the team that made this happen.  The Adhearsion project has exploded in the last year, and many of the people who worked so hard to bring you Adhearsion 2 are actually new to the community within the last year!  A special thanks to Ben Langfeld who has driven much of this development effort and contributed fixes to many bugs and added new functionality in some of our dependency packages in the process of making this happen.  I also want to thank our sponsors, especially Tropo, for not only funding direct development, but helping to evangelize and organize.  Tropo has been a fantastic collaborator throughout Adhearsion’s lifetime.
Now, you might be thinking “all of the above sounds great, but how stable can it really be? Is it webscale?”  The answer is “very stable” and “yes”, respectively.  But I don’t want you to just take my word for it.  A few weeks back, I bet Ben Langfeld a double sawbuck (that is, an Andrew Jackson, a USD $20) that Adhearsion 2 wasn’t ready to take a fully loaded server’s worth of traffic.  And he muttered something about me not keeping the faith, and then took me up on that bet.  So now we’re going to do it live.  In the next couple of weeks we are going to do a live broadcast of a load test, pushing Adhearsion to scale on both Asterisk and PRISM.  We are going to see just how “webscale” it is, and we’re going to be streaming the event live on Ustream so you all can join in the fun.  The loser (hopefully me) will be well and truly prepared to take your jeers and fork over the cash.  Look for an announcement soon for where and when.  It’s about as geeky fun as telephony gets.  I hope you’ll come join us.
In the meantime, go check out Adhearsion 2!

On behalf of the Adhearsion 2 development team, thanks for being you.

Ben Klang
404.475.4841
Mojo Lingo – Voice applications that work like magic
Twitter: @MojoLingo

dahdi timing

Hi,

We heavily use meetme/SLA functionality in Asterisk, and continuously run into issues with dahdi timing. The two errors we get are:

ERROR[6518] res_timing_dahdi.c: Failed to configure DAHDI timing fd for 0 sample timer ticks

WARNING[22024] app_meetme.c: Unable to write frame to channel

Right now, dahdi in our setup uses the software timer (with res_timing_dahdi.so which gives much better results compared to res_timing_timerfd.so). On an idle system (Centos 6, Asterisk 1.8.7, dahdi 2.5), dahdi_test results are pretty good at about %99.99. However, when loaded, the numbers fluctuate between %99.90 and %99.99 which seem to cause the above errors.

We tried Sangoma UT50 as a timing device (it’s USB based), but the test results were worse (between 99.98 – 99.97 when idle, and had big dips when loaded and there is no IRQ sharing).

We are planning to order Digium AEX410 cards hoping they will be an improvement over dahdi_dummy and Sangoma. Does anyone have experience with these cards? Can we get a stable performance from them even under load?

Also (for Digium developers who might be reading this list), any plans for decoupling SLA from dahdi timing in future releases of Asterisk?

Thanks,
Mert