I've been trying to search the archives for older messages, but the
archive at:
http://www.mail-archive.com/[EMAIL PROTECTED]/maillist.html
only seems to go back a few days. Is there another archive somewhere
that goes back farther?
___
I have been having the same problem with 800 numbers. NuFone and
VoicePulse always behave the same (when one can't connect, neither can
the other).
I have so far found no explanation for this. Some other 800 and 877
numbers I can call.
Can you elaborate on this at all?
Thanks!
o I
Ah, I was hoping to find the silver bullet, but no such luck so far. I
have tried every combination of SetCallerID and SetCIDNum in my
extensions.conf, both with and without the |a option, on both services
with no luck still.
When I call myself on our 877 number, I can see that the caller ID
Is there a way to disable the transfer function of the # key? When
calling other services, we often need it to access other menus, other
voicemail, etc.
Does this have anything to do with the T and t options in the Dial string?
Thanks.
___
I got bit by this today and was surprised to see the limit of a measly
100 messages hardcoded into voicemail. Is that right or am I missing
something?
Obviously, this should be moved to voicemail.conf. Does anyone know if
there's a reason why this hasn't been done, or if there's already a
Ah, so in a normal Asterisk world, the messages are supposed to be moved
to another dir.?
In our deviant Asterisk world, the voicemails are never checked through
the phone, only through a custom web interface, so they stay in INBOX
until they're deleted. Thus they collect quickly to over 100
I think international number dialed through voicepulse have to start
with 011... (even if you're located in another countery). I asked them
about that once, and that's what works for me (We've been dialing Spain
and Germany recently, but never Japan)
HTH,
Matt
--__--__--
Message: 4
[EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2879 - 10
msgs
Reply-To: [EMAIL PROTECTED]
In the UK it's 00 then the country code. So a call from the UK to my
phone would be 0013036742575.
Miie
Matt Lawson wrote:
I think international
After 2 weeks on bug #981 (Dropped Channels during dual redirect), I
just posted 2 patches and almost have it fixed. The only problem is,
the patch has a side effect of leaving some zombies. That is, the
zombie channel that is created during the masquerade process doesn't get
hungup. If I
Never mind about the zombies. I fixed 'em real good...
I didn't think I would find the solution so quickly. Thanks anyway.
- Matt
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To
I know having it unplugged from the line will cause this, but it's not.
It's an X101P single port FXO card. Most of the time it works fine but
occasionally wigs out. In this case zttool shows a red alarm. Other
times I call into it and it answers but I just hear a buzzing sound. In
a day
I'm into my 4th or 5th day of working on bug #981. I know that part of
the problem is that the fixup routine is called in chan_sip.c. Well in
there is a line that says p=newchan-pvt-pvt. Problem is, that doesn't
exist in this case.
I see pvt described as private lock but that doesn't mean I
Ours are setup to allow GSM or Speex, and I see that using VoicePulse it
chooses GSM. Don't know the official policy.
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We have an asterisk installation that's on a residential-grade DSL and
its port number (as visible from the outside) keeps changing, every time
it registers. fuser indicates that asterisk is only using port 4569 for
IAX2 (as it should), but when it goes out over the Internet, the port
number
Hmm. Both Voicepulse and Nufone don't seem to be able to dial out 800
numbers. Are 800 numbers treated differently somehow? Or is there a
business reason for disallowing them? It makes the ringing sound but
never connects.
___
Asterisk-Users
That's one of the things that's been on our (1control, I have nothing to
do with Digium) wishlist/to do list that just hasn't gotten done yet.
Currently, video in meetme is not supported. What we experience is the
audio will conference with the other audio streams but the video just
freezes.
No, there is no video output once the call goes into a meetme room.
What I was talking about is a case where you have a regular video call
between 2 video phones, then you try to send them to a conference room.
The audio still works but the softphone's (Linphone in our case)
behavior is to
But this is not to say _you_ can't built a reliable VOIP based
system. Get _two_ providers and set up your dial plan in
extensions.conf to fail over if one service fails to
connect to dial via the next one and finally if both fail
use pstn. your users will see a system the just works.
Now there's
OK, so I answered my own question. Turns out case #2 just goes to
extension 2.
Still trying to figure out the optimum arrangement so I don't have an
inordinate number of extensions. Maybe like this:
1. First outgoing try
2. Second outgoing try
3. Third ougoing try
4. Play a message
I was just about to write the same thing. It says busy. Is is REALLY busy or is something else wrong?
This on the heels of switch-1.nufone.net being missing out of DNS.
We have customers that expect their VOIP to work. Is there anybody that's reliable?
I am having probelms connecting to
Check this out. I recently closed a bug I had written, #495
ExtraChannel in transfer causes crash Now I've been able to reproduce
it, and somewhat narrowed down the culprit. But before I write another
bug report, I wanted to see if anyone else had experienced the following
(or would like to
I have observed this also.
Try downgrading the firmware on the Snom to 1.16x. That usually fixes
it for me. Although that's obviously a workaround and not a true fix.
It is the Snom phone sending the 486 message; I just don't know why.
- Matt
Hello All,=0A=0AWhenever I try calling SNOM
Howdy,
I recently saw something strange with a call between *'s over IAX2.
There are actually 3 *'s involved. The setup is like this:
SIP phone --(ulaw over LAN)-- *1 IAX2 (ulaw over
Internet) -*2(GSM over Internet)
---*3(ulaw over LAN)--
On Thu, 2003-12-18 at 15:40, Kevin Bockman wrote:
Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server.
So, I ran
In an attempt to reduce bandwidth usage, I tried forcing my Asterisk to
use Speex. I did a disallow=all then an allow=speex. The crazy
thing is, it didn't reduce the bandwidth usage at all!
I can do an IAX2 show channels and it shows the call being in format 512
(Speex, right)?
Then I
I'm using Asterisk to do audio as well as H.263 video over SIP.
Actually the video works pretty well but I have trouble with the audio.
I'm wondering if someone can suggest codec/jitter settings or other
tweaks. The system looks like this:
Softphone ---ulaw Asterisk #1 --IAX (usually
I'm trying to put multiple Linphones and Snom 200's into a Meetme room.
With two devices, echo is quite noticeable. With 3 or more it
degenerates into white noise. Which part of the software is responsible
for echo cancellation in a MeetMe room? Is it a setting on the phones
themselves, or
Oops, my bad.
Turns out it was just mixer settings, feeding back through the soundcard. Sorry for the noise.
Message: 14
Date: Wed, 03 Dec 2003 17:43:16 -0500
From: Matt Lawson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Echo cancel in MeetMe?
Reply-To: [EMAIL
I see that when an Asterisk connects to another one via IAX, it seems to
use port 4569 for the first one. But if it has multiple IAX connections
the additional ports seem to be chosen at random.
Is there anyway to predict, or specify which ports or range of ports to
use, for the sake of
to CVS. I would definitely like to check on this
one patch though. Thanks. - Matt -Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Lawson
Sent: Thursday, November 20, 2003 7:03 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Solved! Snom 200
With a recent update to Asterisk CVS, and versions 2.02r and t of the
Snom 200 firmware, I'm getting the Snom phones stuck reporting Busy.:
-- Got SIP response 486 Busy Here back from 10.12.34.248
-- SIP/3064-b07d is busy
They're on-hook, not doing anything. They are registered fine. I
As a follow up to my earlier posting, the problem with the Snom 200 Busy
signal was the firmware! I reverted back to 1.16x and everything's OK.
That made today pretty complicated, since I already had a new kernel and
a new Asterisk build I was trying all at once
Hopefully someone else
I recently updated (fresh checkout) to the newest zaptel and Asterisk.
The one I was using before was a couple of months old.
After updating, my zap channels don't work. They won't pick up incoming
calls or dial out. When I try to dial out I get:
-- Executing Dial(SIP/3064-564c,
Ah ha. That's *almost* got it. It will now load and * will run. The
only big gotcha is it won't pick up or dial out on a POTS line. ztcfg
shows both channels configured OK, as does 'zap show channels.' If I
try to dial out I get:
-- Executing Goto(SIP/3063-74d0, outside|9555|1) in new
I've just finished updating my Asterisk to the CVS version.
Unfortunately, chan_zap won't load anymore.
The hardware has not changed and the config files have not changed. I
can re-install the two packages back and forth. The old one will still
work. The new one won't. I tried updating to
Hello,
I'm trying to compile a brand new CVS Asterisk and running into trouble
with include files. I have an older version of Asterisk that I can
compile (2-3 months old) that I can compile fine, but the new one gives
me this:
make[1]: Leaving directory `/home/matt/asterisk_update/stdtime'
Hi all,
I'm wanting to take an existing call, and transfer both sides of it into
a meetme room (yes I know the phones have a conference ability built-in
but humor me). What seems to happen is I can transfer one half of it
fine, but as soon as I do that the other half hangs up. Do I have to
Most of the time, when someone calls in from the outside on a POTS line,
and possibly over IAX as well, they don't hear any ringing sound while
the internal SIP phones ring. If you call from an inside SIP phone,
even forcing it into the incoming context, you hear the ringing.
The outside
What steps would have to happen, in order to take an already-connected
call and move both parties into a conference room? i.e. do both parties
have to be parked first, or can one or both of them just be immediately
transferred to a MeetMe extension?
How can I loop through something x number of times in the dialplan?
i.e. if I get an invalid extension I want to re-play the menu, but not
forever. Maybe 3 tries or something.
I'm pretty sure that I've seen it before, where you can increment a
variable and do Gotos based on it. But I've
Hi everyone,
I know someone makes a product that's a POTS phone to SIP converter,
where you just plug your POTS phone in one side and the network cable in
the other. Has anyone successfully used any of these with Asterisk, and
if so how expensive were they?
I ask partly out of frustration
Is there a way to determine which channels belong to fxo vs. fxs
devices? I need to write an auto-configuration program that can match
up channel numbers to types.
I have to assume there's an unknown ordering of fxo and fxs cards.
Suggestions? TIA
We have some regular POTS phones connected to our incoming line as well
as the machine that runs Asterisk. Sometimes during an outgoing call
from the POTS phone, the Asterisk will pick up also, and play its menu.
The FXO card is set to fxs_ks signalling; I'm told this might be the
culprit but
When I call in from an outside POTS line to a Zap channel, and the call
ends, it seems like the hangups are very sloppy. I see Asterisk give
the hangup command, but on my phone there's lots of clicks and the line
acts like it's staying open for several seconds, then I hear a phone
ringing
I have discovered something quirky in our Asterisk. If I call in to a
Zap channel (from an outside POTS line), then transfer the call around
several times, I get the above error, after which it will hangup. I
believe Asterisk may issue a SIP CANCEL to the extension it was starting
to dial.
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