Stick the lobby phones into a call group and put your other phones in that
pickup group. Then you can hit *8 to pick up those calls (or, if you have speed
dial/BLF/softkeys, program one of those as *8 for an immediately accessible
button).
In sip.conf for the lobby phones:
callgroup=1
in
Greetings list,
It seems that more and more phones these days are coming with XML
mini-browsers. I'd like to have a go at developing something useful to use on
them, but in all honesty, most of our customers use their phones to make and
take calls and very little else.
So I'm open to
Greetings list,
Wondering if anyone's come across this before.
I've configured a couple of our servers with a privatedundi context to allow
calls to still flow between extensions even if they're registered to different
servers . The DUNDi lookups seem to work fine, evidenced by the following
Greetings list,
Thanks to all who replied to my thread a few days ago SIP devices with packet
loss tolerance. One of the suggestions that came out of that thread was to
replace routers at users' premises with ones that support QoS.
I've used m0n0wall's QoS in the past with reasonable success,
What I was asking is how the traditional telco guys get new
sales/support/consulting business. With IT it's usually a combination of
cold call/networking/word of mouth. I'm hoping that Telco is the same but I
never see any telco guys at networking events so I am thinking they cold
call and
2. Get in good with commercial realtors, they can provide huge leads.
I'll second that. Companies developing office buildings are always a good bet,
as are architects.
Companies providing managed offices are an even better bet - especially if
you're in a position to bill each tenant
An interesting definition of non-commercial discussion you have going
there...
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Greetings list,
Hoping someone might have experience with poorly-performing net connections and
which devices work best over them.
One of our clients has a number of employees that work from home, and are given
a SIP phone to take with them and hook up to their broadband. For the most
part,
can you help me please???
We're in a better position to help if you can post your Zapata.conf
zaptel.conf files for us to take a look.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from
Thanks for all the replies. Answering the points raised in turn:
How did you perform the speed tests?
Generally using thinkbroadband.com's speed test java applet.
On the matter of the BitTorrent factor: did you have the users connect
the phone, and only the phone, to the Internet connection?
It seems that I need the serial number to get a free copy of HPEC... but
unless someone can convince me otherwise, I have a feeling it would just
be easier to shell out the $10 per channel to avoid the downtime and
drive out there.
Not that I'd normally encourage cheating the system, but
Or can anyone suggest something similar - I need a console with about
25-30 buttons/lamps, sourced in the UK ...
I've just had an Aastra 57i w/ LCD sidecar delivered for testing... initial
results look very promising. Might be worth looking into those if you want an
alternative to the
Ah, back in the old days our government privatized the state monopoly
(BT) intact (attitudes and all).
For all their customer service failings at the call centre end, once you do get
an engineer, it's very rare to get a bad one.
Many of them have been in the job for decades and, at a guess,
Greetings list,
A quick question regarding extensions.conf #include behaviour if I may. I'm
sure someone will know the answer off the top of their head...
How does asterisk handle overloading of contexts. For example, say an
extension exists in extensions.conf as follows:
[incoming]
some
I have previously had good success on smaller installations with TDM400P
cards. I now have a UK customer looking for 8-10+ lines and it seems
like a PRI would be most economical + reliable?
Best bet would be to talk to insert telco of preference and ask them what
they recommend. For anything
If there was something useful in ones kettle having an ethernet
connection, it would probably already have it. After all, with NAT'ing
there's no real shortage of IP-addresses. And perhaps we would already
have K2K networks, with K2K proxies etc.
It'd be great if I could get my kettle to
However, my experience hasn't been that VoiP is as reliable
as copper lines and so unless you can tolerate the odd outage once per
month or two then you might want to stick to copper for the main
carrier? Does this match with the experience from others?
Until recently, I'd have agreed
Greetings list,
There have been quite a few posts on the list over the last few months about
using DUNDi to ensure users are always reachable even when logged into
different asterisk boxes (as part of a load balancing cluster).
For example, yesterday, this was in a post: (Olle Johansson)
In
After working with the Grandstream GXP 2000 series phones, I have
decided that I am quite unhappy with their problems, both voice
quality, volume, features and others. For their price now, there are
plenty of phones to choose from as well.
A couple of years ago when we first started
How about using one queue to provide the caller with music and position info,
then delayed dialling on the mobiles:
[queue]
timeout=30
retry=0
joinempty=yes
member = SIP/201
member = SIP/202
member = SIP/203
member = SIP/204
member = Local/mobile1@delaydial
member = Local/mobile2@delaydial
What are peoples experience with the reliability of the
TDM400p. Specifically in the 2 FXO, 2 FXS configuration, which is
the 022 (?) model.
Is this board prone to random failures?
We have quite a few dotted around clients' places to handle emergency calls
(and a few other call types we
Greetings list,
Wondering if some kind soul can help me with an issue with chan_sip segfaulting
as soon as it loads...
Basically, if sip.conf contains any peers with host=dynamic in them, asterisk
won't start. Doing -vvvdddc yields the following:
[chan_sip.so] = (Session Initiation Protocol
Greetings list,
Does anyone have any experiences they'd like to share deploying these phones in
medium-size asterisk setups, e.g. 40+ users? I have a project coming up to
deploy 100 phones over 2 offices and the client rather likes these phones. Are
there any obvious pitfalls/configuration
His vindictive dialer isn't playing while it is listening to rings or
busy signals.
Forgive my ignorance, but what on earth's a vindictive dialler? Is it one
with a strong sense of revenge? :-)
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit
Greetings list,
Quite a few of our users seem to be experiencing poor voice quality when
they're using internet connections over which we have little or no control
(i.e. they're using their own router with no QoS, etc.). Some of these
connections are giving a qualify time within asterisk of
Greetings list,
Some time ago (probably about a year ago now) we disabled IAX jitter
buffering on all our boxes because it was causing issues in a mixed 1.0 and
1.2 environment.
One thing I've noticed over the last few months as more and more clients
have moved from the 512k/1mb/2mb ADSL
Beware that ADSL uses vastly more bandwidth than you expect on small
packets, eg if you are classifying using a cheap router then you
probably need to at least half your claimed bandwidth in order to
make the prioritisation work correctly. I added some (hack) patches
to fix the linux
Greetings list,
We have an issue with call quality at 2 sites where the users (4 Elmeg
IP290s at one site, 2 SPA942s at the other) do not have an asterisk box
on-site. Each site has an 8mb down/448k up ADSL connection and the phones
connect via SIP to an asterisk box in a datacentre using g729.
Eric said:
This should be a FAQ. Set the RTP packet size on the SPAs to .2
instead of .3
Thanks for the suggestion. I've logged into the offending devices and set
both to .2. I'll see how it goes for 48 hours or so.
I've looked at the Elmeg ip290's and they are set to 20ms from factory, so I
The first step of getting the phones to log in as their same
extensions as work is easy and works.
By definition, I guess that automatically logs out their office phones?
Has anyone tried anything like this? I would like the phones to
regrab their spot once the softphone is logged out.
Wondering if anyone on here can help with a niggling issue: One of our
extensions is unable to make attended transfers at all.
The phone in question is an Elmeg ip290, and works fine for direct
transfers. However, on attempting to make an attended transfer, the first
leg succeeds (the inbound
the change of Telus' (the
ILEC) customer service system entirely to speech recognition. It
actually works really, really well I've never been able to screw it
up
What happens if you yell I just want to talk to a human being! really
loudly at it? ;-)
Regards,
Chris
--
C.M. Bagnall,
Greetings list,
Does anyone know if there's a maintained 1.4 ebuild for Gentoo? Even with
the ~amd64 keyword, latest in the official Portage repository is 1.2.13.
Thanks in advance.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons
I need to ring a group of 8 phones, but not if they are already on
another call. How can I determine which of those 8 phones are busy so
I only ring the others?
I've done this in the past by disabling call waiting on the phones and put
all 8 phones into a ringall queue. Then, when you
Greetings list,
Has anyone done any research into call routing and transcoding performance
using a Via Epia based platform?
We have a client with a box in a datacentre with 2 PRIs going into the
machine. We've moved most of their asterisk handling into another more cost
effective datacentre, but
For HFC cards in the UK, check out Solwise - www.solwise.co.uk
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons
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asterisk-users
I just bought a grand stream 2000. It appears that it will
not dial any number with a leading * (*70,*71)
So I can not dial any of my Apps in *
What firmware version are you running? We have plenty of GXP2000s in
clients' premises with plenty of numbers beginning * without any problems.
Greetings all,
Not specifically an asterisk query, but a couple of transfer queries that
I'm sure are obvious to folks who use these phones all the time:
1) how does one do a blind transfer? When a call is answered and one hits
the transfer button, followed by an extension, one has to wait for
Greetings list,
Before I go and write something from scratch, are there any kind souls here
who already have a nice code fragment that works our charging for calls
split across charging zones?
There are essentially 4 possibilities for a call:
1) call is completely within one zone, so it's nice
I think the issue for many people here is not the cost of the licence
itself, but the very frustrating lockdown to specific pieces of hardware
without any real reason.
I say without any real reason because anyone who doesn't care about the
licencing of g729 has an easy alternative in the form of
Greetings folks,
I have a client who wants to divert his main office number to a call
answering service when the phones in his office aren't answered within a few
seconds. Calls to the main number go into a queue which then rings all the
phones in the office.
I've changed the timeout in
I downloaded the SLIN and I have a couple of remarks.
1) you can't use the SLIN directly on a non-intel machine -
you may have to byte swap it first (took me a while to work
out why I just got pulse modulated static on my NSLU2 home
asterisk! (armv5teb) )
Can you explain in a bit more
I've got a HFC ISDN card that I'm using with chan_misdn and
it basically behaves like crap. Echo is waaay worst then echo
I get TDM400 card, sound is choppy (there other side is
allays complaining about sound
interruptions) and to top it all it detects fake DTMF's all the time.
Is this a
Is there anyway in which I can successfully receive incoming
calls from my Voip-Talk.org numbers (an 0845 number) without
the static IP?
Can you login to the voiptalk control panel and change the numbers to point
at your current IP ?
I've tried various options with SIP IAX2, but it would
'recognize'? The phone cannot know that the external IP has
been changed, unless it is using a STUN server and
periodically re-doing the STUN queries (which I doubt any phones do).
Thanks for clearing up my misunderstanding as to the point of STUN. :-) I
thought the phone would query the
Greetings list,
I'm coming across an issue with some of the GXP-2000 phones we have out in
the wild at clients' employees' homes. In most cases they're behind consumer
ADSL NAT routers on a dynamic IP from their ISP.
In a nutshell, the phone is unable to be called unless it's restarted first,
Isn't a TE110P a PRI card? Are you sure that's the right model number for an
analogue interface card?
For our sites with BT lines, we have them configured as follows (I've
extracted the settings I think might be relevant):
usecallerid=no
hidecallerid=no
callwaiting=no
callwaitingcallerid=no
I've no experience with the A200, so I can't really pass judgement on it. We
have 2 sites with TDM400P cards currently deployed (one with 3 FXO, the
other with 1 FXO and 3 FXS). The site in Northampton with 3 FXO has been an
absolute nightmare over the last 9 months the system's been in place.
How does a large file transfer like your excel spreadsheet
example, affect communication between an Asterisk server and
SIP phone? The only possible configuration I can think of
that would cause a problem is if the client PC is sharing the
same eternet cable and therefore the same
...or if your
asterisk server is also a file server (which should never be
done)
I know I'm attracting flames for disagreeing, but sometimes when you're
dealing with small business customers there simply isn't the budget to have
separate machines for doing x, y and z, and often one finds the
Can this be done? I havent seen to much of this on the
mailing list, im guessing each server would talk to the main
* server via a IAX trunk or a SIP peer. Also one other key
point would then be to keep the voicemail for each office on
its local * server instead of having it go to the
We could implement this in SIP, by forcing an outbound
subscription, but if all the servers are Asterisk servers
there has to be more simple ways to solve this as well as
cross-server voicemail notification.
Could you elaborate on that please? I'm almost certain to come across the
For handling outbound calls, your easiest approach is to have each business'
phones default to different contexts, so you might have the dialplan
arranged as follows:
[in-pstn]
exten = number,1,dosomething
exten = number,1,dosomething
exten = number,1,dosomething
etc.
[business1]
; internal
Greetings all,
Has anyone managed to get dialplan status hints working across multiple
servers? I've separated a load of SIP users out across 2 servers today, but
it'd be useful if they could still see each others' status.
I've replaced the various hint lines for the sip devices now on another
I hear some
people praising the GXP2000 phones and I gotta wonder what
they are smokin (regardless of firmware revison) so I just
don't know who to believe anymore.
As one of those who's praised the GXP2000, I feel I should just add that
it's all relative *to the price point*. The GXP2000
I agree with most of Raymond's other points, but I have to take issue with
this one:
1) If it doesn't support PoE I won't implement it. Support
phones with wall-warts or bricks is just a added hassle and
adds TCO as most end up being replaced once or twice during
the lifetime of the phone
What's the benefit of using stund vs nat=yes in your sip.conf
for that device? I haven't had any issues behind firewalls
when I enable that option, and no ports are needed to be opened.
For some strange reason, even with nat=yes sometimes when a user's IP
changes, the phone doesn't realise
PS A central resource of various Voip terminators and the
quality of routes to/from various ISP's would be a great
boon. Is there such a thing?
When we've added asterisk servers (in datacentres) to our collection, one of
the things I've always asked the datacentre to provide is a traceroute
£40! That would be a cheap and nasty switch with no prospect
of any management. A managed switch is worth its weight in
gold, /especially/ when you have to look after things remotely.
How does one justify the extra cost of a managed switch for an office of no
more than 5-10 users with
What I'm trying to do is accepting a call from pstn, put it
into a queue, while callee is waiting contact some numbers
till one responds, then bridge the two calls.
What I can't manage is jump to next dialplan command soon
after callee enters the queue in order to call other numbers.
I've
As I recall from various firmware versions on the spa3k,
incoming pstn calls are forwarded to asterisk meaning the
incoming call is answered and then forwarded. Later versions
did something a little different.
I can definitely confirm that the SPA3000 here at home forwards the call to
Greetings all,
I'm trying to improve the codec selection on a few of the asterisk boxes we
have to keep the g729 licences free for calls from ATAs that don't support
anything apart from g711 and g729. GSM seems to offer noticably inferior
call quality (at least when using a softphone + decent
The GXP2000 firmware is not bad for features and ease of use
but still buggy. The hardware is junk to be quite honest and
I don't think firmware will ever fix that. The Aastra 9133i
hardware is 10x better.
I have a few of both here at the moment, and I'm not sure I'd agree with
that. The
I know it's still beta, but don't use the latest firmware in
production unless you can live with an empty display after
transferring a call.
Only a reboot of the phone will give you text on the display again.
I tested and confirmed this with 5 phones.
What firmware are you running? I've
I do not even know which brands/models to consider that are
out there. Given that we are in the US, and want to use BRI
to improve sound quality (no echo, no static), what would be
some good cards to look at? I hear a lot about BRIStuff,
which I think is used on the Junghanns cards (like
My 5 cents worth is if you use Bristuff stable you must use
Asterisk-1.0.10 ( Old version ) If you use Bristuff 3PRE1l
you will have problems with FXO cards as I did.
Bristuff3PRE1l is not Stable use at own risk!!!
Can't speak for anyone else, but we have 2 sites running HFC cards with
I'm helping out with a political campaign and would like to use
asterisk to blast out about 200,000 calls with a short
message from the candidate.
I can't speak for anyone else, but I'd find it very difficult ethically to
be involved in this, even assuming it's legal.
Check very carefully
Have you looked at Wildfire (was Jive Messenger) with the Asterisk-IM
plugin? It seems to work fairly well in my experience. I have it running
here at home and also on one client's network. The XMPP client they provide
(Spark) is a bit primitive, but something like Trillian also supports the
Hello all,
I've started implementing iLBC on some of the ATAs we have floating around
clients' homes, but I'm coming against this error message with most of them:
codec_ilbc.c: Huh? An ilbc frame that isn't a multiple of 50 bytes long
from RTP (38)?
The ATAs in question are various
Greetings all,
I'd like to start implementing a private DUNDi peering group between one of
our asterisk servers hosted at a datacentre and the various asterisk boxes
sitting at clients' premises.
On most of the clients' boxes the dialplan will have an [in-pstn] section
containing the various
Would a call coming in on the pstn line be answered by the
ATA or just get passed through to the * server (depending on
dialplan) to handle?
Either. It's your choice. I have an SPA3000 here at home working in the way
you describe. When a call comes in on the SPA3000 it's forwarded (without
The ATA will answer the POTS line, therefore the caller will
be charged as soon as the ATA has tried to grab caller id and
picked up the line (usually around two rings).
This is incorrect. Whilst the SPA3000 *can* work this way if you wish, it
doesn't have to.
Regards,
Chris
--
C.M.
You need to get BT to agree and allocate or port the numbers.
You need to agree how many digits BT will pass on to you
(probably 1925838395 but possibly just the last 2)
I don't know the number of digits that BT pass through on a PRI, but on a
set of BRIs with a range of DDIs, they're passing
i'm planning to migrate a callcenter to asterisk and VOIP,
the call center can have up to 25 cuncurrents agents logged in.
Can a normal server with
1 GB ram
100 GB HDD
Pentium 4 3.6 Ghz CPU
Ethernet 10/100/1000
One of our clients has a similar sized setup running on an Athlon64 2800+
these are the lables on the softkeys when having a phone call:
Holt / EndCall / Confrn / more
press more and you get
Transfer / / BlndXfr / more
Is this specific to the SIP firmware? I'm using chan_sccp with a few 7960s
and Transfer is definitely on one of the initial softkeys when
I don't think there's any way of busying a line at your end unless it's
genuinely busy. The best folks to talk to about that would be your telco.
You could fudge it somewhat by artifically leaving the line offhook (I guess
this is possible in asterisk somehow - possibly by using a dialplan
The issue is that their dialogic solution can read callerid
from incoming calls, even if the callerid is blocked.
I don't know what the laws on such things are where you're located, but you
might want to check into the legality of actually doing that.
Regards,
Chris
--
C.M. Bagnall,
The developer has indicated that a revised version of the
PSGW (http://www.rsdevs.com/) code will be available for sale
shortly with the changes.
Has the developer indicated to you whether this would be a free upgrade for
existing clients or whether additional payments would be expected?
I played with FXO on the HT488 a bit, but didn't have a whole
lot of luck. We had a bit of a problem with echo, but more
seriously the thing kept getting itself into a variety of
wedged states: sometimes it would lock up altogether (usually
with its button lit up), and sometimes it would
The only significant feature that the SPAs seems to be
missing compared to the HTs is the Early Dial thing (where
it sends each digit to Asterisk until it gets something other
than a 484 response back).
Has anyone ever gotten that working? I've tried it on every Granstream
device I've had
I am new to asterisk and am looking for a voip provider
that supports asterisk. I am aware that their are several
vendors to choose from. Any opinions on the best one?
I think more information will be needed before someone can give you a useful
reply.
Things you might want to consider:
I installed one and works fine but of course when I try
to make the second call it says no lines are available
That's weird. I was under the impression the non-Digium ones didn't care how
many lines were in use, as there was no monitoring of such things in there.
Regards,
Chris
--
Hi I'm looking for a pinout for the above. Note this has
what i'd call
RJ45 sockets (or someone smart can correct me). I need to
plug into BT (rj13?).
Are you sure the TDM400 has RJ45 sockets? The pair I've got here have RJ12
sockets.
I assume with the mention of BT, you're in the UK.
I am venturing down this path soon, trying to decide if I
will be okay sticking with 2 x Fritz! PCI cards or if I
should really get a different card... 4 port ISDN or something...
Chan_Capi with 2 Fritz cards I'm told requires code
modifications to work etc...?
One of our clients is
Don't know if anyone's got experiences on this they'd be able to share...
I'm trying to obtain numbers for Kenya/Nigeria, but I'm struggling to find a
company selling them. There's one for Nigeria listed on didx.com, but
$22/month seems a little steep. Has anyone had any luck getting DIDs for
I am trying to construct a macro for long distance dialling. I have
two internet feeds, I have all routes including Teliax on
Internet A
and a static route to Voxee on Internet B.
Here's an AEL macro I use on our boxes. Modify for your needs.
// dial a number with a range of routing
Successfully got the adapters to allow the BT phones to ring
on lines coming out of a TDM.. but now my latest
problem is echo.
Suggestions / Experiences in UK appreciated
Most of our clients with BT lines tend to have ISDN BRIs, but we do have one
in Northampton running 3 analogue
Use Krone cable and a genuine Krone tool It isnt the
cheapest, but it is the best.
I concur with you on the genuine Krone tool, but I'm no fan of their patch
panels.
I find STP patch panels are much nicer to work with (even where STP isn't a
requirement).
On cable, we tend to use Belden or
I've got the same issue than you. Have you solved your problem ?
I enabled Enhanced Real Time Clock Support in the kernel config,
recompiled the kernel, then recompiled Zaptel.
I found the rtc: lost some interrupts at 1024Hz messages seem to be
related to rebuilding arrays on my RAID5
I am not sure that it is a problem more so an annoyance. If
someone dials my extension number or external DDI while I am
already in a call rather than skipping to the next priority
in the dial plan for example voicemail the line continues to
ring and while in a call I can hear the phone
pain to configure) have 4 ring types. I am guessing that I would
need to figure out how to tell this particular phone to use a
different ring tone unless there is a way to send a
stutter type ring to the phones.
Has anyone found a solution to this?
I did a similar thing for a
non-commercial is a misnomer, the patent may still apply for
your usage, then again it may not. The libraries that are
used are intels IPP which are free for non-commercial
non-distribution purposes, if you want to distribute you have
to pay intel money, but that gives you the core from
I also didnt comment on whether or not anyone can prove that
you do have licenses, even if they know you use the codecs.
Because to rely on that would be dubious at best, shut you
down at worst.
Out of curiosity, I wonder what one's legal position would be if one bought
the appropriate
So,
if
${CALLERIDNUM}=0123456789
Then
${CALLERIDNUM:3} returns 3456789
${CALLERDINUM::3} returns 012
${CALLERIDNUM:3:3} returns 345
But this do not work anymore in 1.2.1, and if I do not found
solution for this I will downgrade to 1.0.9
Have you tried ${CALLERID(number)::3} ? I have a
I know there are numbers
provided by other providers in UK
If you happen to know of any, please feel free to post them. :-)
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons
___
I have a GSM box, which needs to connect to a analogue phone
line. I've plugged the GSM box to a Grandstream ATA (386).
The HT386 only provides a lifeline on the line port - it's not a true FXO
in that it doesn't register with Asterisk as a separate entity. The one you
want to do what you're
I think it would be biggest is in consulting. The people that
refuse or cant to pay for call manager or Avaya's one.
That's certainly been our experience over the last 9 months or so we've been
involved with Asterisk. The bigger companies don't seem particularly
interested (or if they are,
I have it working both here and at a client's place. I'm using Trillian Pro
as the IM client here, and Spark on the client's computers.
Seems to work pretty well. I seem to remember the steps I followed were:
1) Set up a manager account in asterisk (manager.conf)
2) Install Jive Messenger Server
I don't get it. What is the advantage of using a GSM gateway?
VOIP calls are pretty inexpensive as they are now.
It largely depends on the country you're calling. Here in the UK, calls to
mobiles are maintained at an artificially high rate because the terminating
network (the mobile networks)
Has anyone using a GSM gateway incorporated some time monitors into their
dialplan?
For example, if a SIM card has 400 inclusive minutes to any network in a
month, I want to make sure that Asterisk doesn't go beyond that unless it's
for calls to the same network (once you're out of inclusive
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