Bartosz Piec wrote:
How to set calls forwarding only to numbers that are available in user's
context (so if he has only locals calls he cannot set calls forwarding
for mobile phones)?
When the user sets the forwarding number, store the user's context in
the DB along with the forwarding
Gary Richardson wrote:
I'm running 1.2.4 and just about every call is cut short. I'm using
Cisco IP phones as end points. All the outbound calls are routed via SIP
through a PRI line attached to a Cisco 2811..
For me, all incoming/outgoing calls arrive/leave via IAX2/ilbc and all
the local
Gary Richardson wrote:
Now it seems that if I'm really loud on a call, MixMonitor stops
recording. The wav file stops growing. The log says nothing. When you
hang up the call, MixMonitor reports that it is exiting, even though
it hasn't been recording since that loud noise.
Has anyone
Sam Tam wrote:
Do anyone know who can provide some cheap PH routes/.’
I've been looking myself. Cheapest DIDs in Metro Manila I've seen are
$27.50/month; cheapest termination to same (non-mobile) from US I've
seen is $0.23/minute.
Expensive chismis :-)
-Johnathan
Michael Collins wrote:
I had never sat down to write an AGI script before - I hadn't needed one
- but I thought, How hard can it be? Ugh. The Asterisk::AGI module
is very handy, and I highly recommend it. I've only written one AGI
script in my life (up to now) but I've written 10's of
C F wrote:
Can you explain this?
What country?
In this case it's not asterisk but the telco that has to do the Answer.
To every mobile? or just that provider?
I too have seen something similar in the past. When calling Verizon
(408-489) numbers, when there is no answer and it rolls over to
Roger Lewau wrote:
If the voicemailbox contains messages the voicemail application exits
with a non-zero status either when reading the number of messages or
when selecting 1 for listening to new messages.
Is it possible the permissions for the sounds directory or individual
files within have
btb wrote:
[7508] ;ipkall
type = peer
dtmfmode = rfc2833
context = remote
callerid = ipkall incoming 7508
nat = no
You've configured this entry as a peer, which is for dialing out, versus
as a user, which is for incoming calls. Solution is to change to
'type=user'.
If you really need a
Douglas Garstang wrote:
Yes, programming the dialplan is akin to programming assembler.
Too funny. But true.
The first time I did a 'show dialplan' after trying out AEL, I felt like
I was seeing an assembler dump of C++ :-)
-Johnathan
___
Andres wrote:
There is no username in the above To header. Check your DIAL command
because something is wrong here. Thats why you get a 404. The SPA
can't match the username.
Yes. I had not reverted to an early enough commit on the configuration
files and the usernames were still missing
I don't know what's changed, but four SPA841s and a SPA3000 are no
longer answering when they get an inbound call from *.
This has been a working configuration for weeks. I *have* been fiddling
with the server config; however, the configuration is under version
control and I've reverted
Bastian Schern wrote:
is it possible to limit the maximum runtime of the command echo?
Use the AbsoluteTimeout application in your dialplan preceding the Echo
application.
http://voip-info.org/tiki-index.php?page=Asterisk%20AbsoluteTimeout
-Johnathan
I can call my Broadvoice DID from a outbound caller-id blocked phone,
and BV happily delivers the CID to Asterisk (and then on to my IP phone
display.) I've tested with the *67 prefix from a PSTN phone to make
sure it was supposed to be blocked. The number is always correct, but
sometimes
If I run sox (on Linux), just specifying the input and output files by
with the right extensions, it will convert a raw gsm file to a wav
format file while retaining the gsm compression:
sox vm-youhave.gsm vm-youhave.wav
This is without any additional options. The output file is playable on
Robert Goodyear wrote:
So: knowing that the X11 window GUI is a resource hog, is it appropriate
to use the GUI to install and configure various components, then set
RUNLEVEL to 3 once all is nicely set up and running cleanly? Would this
give the same effect as doing a minimal install or is
Robert Goodyear wrote:
Noted. To clarify, will dropping back to runlevel 3 still ensure a
smaller set of processes that would be as non-intrusive as if I had
installed Linux with console/command line support only or would there
still be stuff hanging around that's inextricably there because I
John Novack wrote:
...along with detection of stutter dial tone on analog lines.
This would be somewhat useful. When there is stutter for whatever
reason, outbound dialing must be delayed a couple seconds. Some times
my PSTN line would roll-over to the telco voicemail before * would
answer,
Mike-Olumide, Johnson wrote:
I get fascinated when I dial someone and get an IVR play for accounts
department press 1, for sales, press 2 or hold the line for an operator
and then have MOH play before the line is picked up at the desired extesion.
You'll find a simple example of how to
Mike Clark wrote:
However, outbound calls are hit or miss. Sometimes they work fine and
other times we get a you must first dial a 1 or 0 message back from
telco when dialing out standard POTS lines.
Did you get this working yet?
-Johnathan
___
Mike Clark wrote:
We are installing a number of systems with 2 TDM04B cards. Have done all
the isolation to unique IRQs, etc. All inbound calls seem to work fine.
However, outbound calls are hit or miss. Sometimes they work fine and
other times we get a you must first dial a 1 or 0 message back
Jeffrey Starin wrote:
Jonathan! You don't know how much that simple explanation has helped me
understand Asterisk. Well done. Well said. And to the point clearly.
I would hope this could find it's way onto the Asterisk Wiki and be the
*first* thing someone reads when looking at the
Ronald Wiplinger wrote:
I cannot email them, I cannot call them, I do not get an answer, but the
credit card is still charged, although NO phone calls are possible
anymore, ...
Hmm. I called them twice yesterday to ask questions, the queue wait was
less than a minute in both cases. First time
Kerry Garrison wrote:
PSTN line 1 - Parent's line
PSTN line 2 - Kids' line
PSTN line 3 - Bussiness line
VOIP Provider 1 - Broadvoice
VOIP Provider 2 - VOIPJet
A call into Line 1 answers For Kerry press 1, For Karen press 2
A call into Line 2 answers For Taylor press 1, For Chris press 2
A call
Jeffrey Starin wrote:
I have a little confusion about the general settings (other than the
register values) in the SIP
General area.
[snip]
However, I'm confused as to the purpose of the
general settings -- to what or which connection do they apply? Since
the context suggested for the general
With the recent service outage at Broadvoice, there has been a lot of
discussion here, on broadband reports, Voxilla, etc., regarding whether
VOIP is mature, or ready for the masses, etc.
One particular point I've seen repeated, and with which I agree:
we're willing to deal with less than five
JD wrote:
Inbound
calling has been down for 2 days.
Just FYI, mine is back up (408-903) as of about five hours ago.
I did just speak with a (Broadvoice) support tech on an entirely
unrelated matter (40 min. hold time!), mentioned mine was working, and
he seemed to think things were coming back
Kumara Jayaweera wrote:
Sorry for the numerous postings. but How could I slim my Asterisk PBX.
Really I don't need such modules like Ex. chan_modem.so. Becouse, I don't
have any special hardware. Please, could I hope your various suggetions in
this regards. brief me your idea.
See the Wiki under
Josiah Bryan wrote:
Why not just NFS mount the /var/spool/asterisk/voicemail directory from a
central server? That way, all servers share the same spool and the MWI will
get reflected on all servers.
Does * use any form of locking to maintain the integrity of the sequence
number for voicemails
Jason Brown wrote:
Trust me though, I promise, 1 central VM store does work and work well in an asterisk environment.
But I haven't seen addressed the issue of two or more servers sharing
this central VM store, when running the Voicemail application. Sure,
MWI should have no problems in the
Brian Watters wrote:
Ok .. It appears we have outbound calls working well via our SIP trunk and
BroadVoice .. However inbound just plain does not work, What happens is when
you call our BroadVoice number it tells you the person you are calling is
not available .. Never makes it to our Asterisk? ..
Craig Simon wrote:
Looking for 100 in from-broadvoice
It looks like * is searching for extension 100 in the 'from-broadvoice'
context, not finding it, and sending a 404 back.
First, you can create a extension 100 in that context in your dialplan,
then see if that allows the call to come
Craig Simon wrote:
The 100 is an extension I created for my softphone to log into.
* is tricky with terminology.
You didn't create an extension 100, you created a SIP peer/user named
100, which the softphone connects as.
Extensions (that are within contexts) are lists of commands that *
will
Craig Simon wrote:
Thank you, thank you, thank you! That was it! Thanks alot for the
description, it made the call flow much easier to understand.
You're welcome. I only discovered Asterisk about a month ago myself,
and understand first hand how difficult it can be for the uninitiated.
Dalon Westergreen wrote:
BV allows unlimited incoming, and up to 3 outgoing. My understanding
is that they intend to charge for more 3 outgoing, but have not done
so at this time.
This is good to hear--do you have anything from BV that documents this?
Also, being relatively new to *, I don't know
Bob Goddard wrote:
The apparent packet loss you are seeing may be just fine tuning
of the routers in question.
This is the conclusion I came to as well; however, with the way
PingPlotter works the router is not sending ICMP unreachables but rather
ICMP TTL expired responses. In any case, the
Rich Adamson wrote:
In other words, as the ttl value is increased and additional icmps
are sent, you might see what you believe is congestion, but you still
don't have any clue as to whether hop #2, #5, or #10 actually was
involved with that congestion.
Sure. But there is a way around this.
The
Johnathan Corgan wrote:
First off, I have Sprint Broadband Direct internet service, a fixed
wireless setup with a 2-5 Mbps downlink and a terrible 128 kbps uplink.
So I know I'm in for trouble anyway.
The broadvoice edge router (63.251.209.126, their lax site) is another
11 hops away. One hop
Joseph Gutowski wrote:
I installed PingPlotter, switched to UDP just to be the same as you,
and ran it against sip.broadvoice.com. Absolutley no problems, no
packet loss at all.
Well, that's good to hear.
I then used the 63.251.209.126 that you posted, and it was awful (at
least it appears awful).
Robert Terzi wrote:
The best tool I've found for monitoring connections, routes, congestion,
is called PingPlotter. http://pingplotter.com/ It's a shareware
visual traceroute. It continually graphs the traceroute style
responses. There is a scrollable timeline to view how things change.
You
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