* You are viewing the archive for June, 2012

Gigaset in the USA

Does anyone know if Gigaset is for sale in the USA? Based on my assessment of phones and features, i would like to try the N300IP base along with C610H phones.

I can only find the handsets on ebay, no retailers in USA. And I suspect they are using European frequencies.

Park function and billsec

When a call comes in and is parked using the Park() command it appears that
the billing seconds do not include the time while the caller was parked. Is
there a simple way to correct this with an option or setting in asterisk.
This is not acceptable as we are loosing min when callers park calls.

I am using 1.8.x release currently.

Thanks
zktech

Intro to DECT vs IP

We’ve deoplyed a number of pure VoIP wireless (wifi & proprietary) phones, but not dect.

Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect?

Can you push configuration info to individual phones? (Are they individually addressible / configurable through SIP) etc?

Thanks

VoIP Company looking for Asterisk/VoIP Engineer

Hi,

I work for a VoIP provider in Southern California. We are looking
for someone very knowledgeable in Asterisk/VoIP to help work on the following:

- Maintenance of current Asterisk servers, updating Asterisk, monitoring
load, and other sysadmin tasks
- Devise and implement scalability strategies so that adding additional
capacity is easy and does not compromise anything about the current system
- Troubleshooting call quality issuses through our network (jitter,
audio dropouts..)

Candidates should have the following experience:
- Minimum 3 years working with VoIP/Asterisk
- Have worked in an environment with a significant number of phones (>500)
- Experience working with Cisco networking devices – QoS knowledge is
a huge plus.

Having experience with VoIP over carrier-class wireless links is a
definite plus.

This is a part-time contractor position. We are located in Southern California,
and while having someone local would be ideal, telecommuting is an option.
Hourly rate DOE.

Please email all resumes directly to me at jlamanna@gmail.com

Thank you.

Dahdi Dropping Calls

Hi Guys

Has anyone seen Dahdi dropping incoming calls with Hangup cause 27?
It only drops whilst we are on the phone?
Its not every single call
Any ideas?

PRI trunk between Asterisk servers does not work.

In article <4FECCD0C.1020000@fivecats.org>,
James Sharp wrote:
> On 6/28/2012 3:53 PM, Ernie Dunbar wrote:
> > We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
> > Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen),
> > and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our
> > PRI to the PSTN and we hope will allow us to failover to other Asterisk
> > servers (ie, Voip2 and Voip3). Voip2 is our current production server,
> > and Voip3 is being turned into our next production server.
> >
> > We’re trying to build a PRI trunk between Voip1 and Voip3. Curiously
> > enough, we’ve already done this between Voip1 and Voip2, so one would
> > think that the same configuration would work between Voip1 and Voip3 as
> > well. However, it hasn’t gone so smoothly. If you’re wondering why we
> > don’t just use SIP trunking between these servers, it’s because faxes
> > are not reliable over SIP trunks. I am open to suggestions however.
> >
> > At any rate, the PRI trunk between Voip1 and Voip3 isn’t working, and
> > that’s my current problem.
> >
> > – I have built a T1 crossover cable, and it’s plugged in between Span 3
> > on Voip1, and Span 1 on Voip3.
> > – I have a green light on both PRI cards for the appropriate spans.
> > – Both servers detect their cards on boot.
> > – DAHDI is installed on both servers, and all diagnostics are good, ie.
> > dahdi_test returns good results, dahdi_tool shows that the alarms are
> > OK, and executing ‘dahdi show status’ on the Asterisk console shows the
> > same.
> >
> > The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like
> > this:
> >
> > ; Span 3: TE4/0/4 “T4XXP (PCI) Card 0 Span 4″
> > group=3
> > context=default
> > switchtype = national
> > signalling = pri_net
> > channel => 49-71
> > group = 63
> >
> > ; Span 4: TE4/0/4 “T4XXP (PCI) Card 0 Span 4″
> > group=4
> > context=default
> > switchtype = national
> > signalling = pri_net
> > channel => 73-95
> > context = default
> > group = 63
> >
> > Span 4 goes to Voip2, which has a working PRI trunk.
> >
> > The chan_dahdi configuration for Voip3 looks like this:
> >
> > group=1
> > signalling=pri_cpe
> > switchtype=national
> > context=local
> > channel=>1-23
> > dchannel=>24
> > ;channel=25-47,49-71,73-95
> > rxgain=0
> > txgain=0
> > busydetect=yes
> > busycount=5
> >
> > resetinterval=1800
> >
> > I have a test DID, the dialplan for which on Voip1 looks like this:
> >
> > exten => 604484XXXX,1,Dial(DAHDI/g3/604482YYYY)
> >
> > But when I call 604484XXXX from my cell phone, I get no output on the
> > Asterisk console on Voip3, and this output on Voip1:
> >
> >
> > — Executing [604484XXXX@local:1] Dial(“DAHDI/5-1″,
> > “DAHDI/g3/604482XXXX”) in new stack
> > [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable
> > to create channel of type ‘DAHDI’ (cause 34 – Circuit/channel congestion)
> > == Everyone is busy/congested at this time (1:0/1/0)
> > == Auto fallthrough, channel ‘DAHDI/5-1′ status is ‘CONGESTION’
> > — Accepting call from ’778839ZZZZ’ to ’604484XXXX’ on channel 0/5,
> > span 1
> >
> > I’ve also tried connecting span 3 to one of the other ports on Voip2
> > with the same configuration, and I get the same results. I’ve run
> > loopback tests on the TE110P and tested the cable thoroughly.
> >
> > Any input on this problem is greatly appreciated.
>
>
> You’ve got the spans configured as “group = 63″ but you’re trying to
> dial out on group 3 (DAHDI/g3 rather than DAHDI/g63).

No, the group=63 lines are actually redundant. It is the settings *above*
each channel=> line that get applied to the channels when they are created.

To the OP: what does “pri show span 3″ give you on Voip1?

It might be useful to see the complete chan_dahdi.conf from Voip1.
To save space, you can list it without comments like this:

# grep -v ‘^;’ /etc/asterisk/chan_dahdi.conf

Cheers
Tony