On Feb 23, 2008, at 06:52 , Yehavi Bourvine +972-8-9489444 wrote:
Hello,
I've one nokia E65 that works very well with my asterisk box.
The people here don't let me even try it as they are afraid it will
consume the
battery more than when it is used the usual way. Is this true?
Yes,
Fons van der Beek wrote:
I've overwritten the indications.conf with the one from the
sourcecode, stil no luck
Perhaps somebody knows what the correct value for indications.conf is
when using the dutch xs4all as sip carrier??
A simple way for you to test your indications.conf as far as the
Hi,
Is it possible to setup SIP peers with Asterisk Realtime from multiple
databases?
Thanks in advance.
Ash
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shadowym wrote:
I guess someone has to say it.
Have you considered Aastra?
You can argue about quality/features/functionality but I have set up
both and the Aastra are definitely easier to configure and they reboot
quicker.
Nobody ever complains about the quality of sound or
I guess we are back to the fundamental problem: no asterisk generated
sounds on the external call
After implementing the described test for indications.conf
The CLI outputted:
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/0475769XXX-095a8488, ) in new
stack
-- Executing [EMAIL PROTECTED]:2]
While the call is progressing
sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Format
Hold Last Message
82.101.62.XX 0475769XXX 14151-EX-29 00101/703757593 0x4
(ulaw) No Rx: ACK
82.101.62.XX 0475769XXX 6ec6f62d57d 00103/0 0x0
Jens Vagelpohl wrote:
On Feb 23, 2008, at 06:52 , Yehavi Bourvine +972-8-9489444 wrote:
Hello,
I've one nokia E65 that works very well with my asterisk box.
The people here don't let me even try it as they are afraid it will
consume the
battery more than when it is used the usual way.
Vieri wrote:
What I would like to do is have two identical *
servers which accept registrations of sip extensions
4000-4999.
If I define a rrDNS or LinuxHA then I should have
load-balanced registrations.
However, say ext. 4001 is registered on *1 and 4002 is
registered on *2, if 4001
Fons van der Beek wrote:
After implementing the described test for indications.conf
The CLI outputted:
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/0475769XXX-095a8488, ) in
new stack
-- Executing [EMAIL PROTECTED]:2] PlayTones(SIP/0475769XXX-095a8488,
ring) in new stack
--
T.38 is a codec in exactly the same way that GSM or G.729 is a codec, so
yes it /can/ be used at the same time as any other codec - just that
only /one/ codec will be used at a time. What often happens is that the
call will initially be established with a codec such as G.729 or G.711a,
but once
I posted the same question on asterisk-biz mailing list but didn't have much
response. So I am posting it here now.
I need a good, reliable and stable DID provider for USA, Canada and Europe.
I prefer to have fixed monthly rates for incoming and outgoing calls and not
per minute charges.
Will the built-in T.38 support eliminate the need for spandsp? I'm curious
how this will affect iaxmodem.
MD
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis
Sent: Saturday, February 23, 2008 7:12 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] FXO
Tnx for your support Trevor!!
cat /etc/asterisk/indications.conf | grep country=
country=nl ; default location
show indications nl
Country Indication PlayList
=
nl ringcadence 1000,4000
nl dial425
nl busy
On Fri, 22 Feb 2008 19:44 +0200, Yehavi Bourvine +972-8-9489444
[EMAIL PROTECTED] wrote:
When a call arrives I check whether the REGSERVER coloumn is the same as
the
local server or not. If not, then there are two options:
- Pass the call via IAX to the other servers; this makes both server
Nitesh Divecha wrote:
Thanks Doug,
I tried that but it didn't work either... As per Wiki
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail it
I think Trevor is correct. If there is a temp greeting recorded, this
will be played instead of the busy. Remove it and you
Hello everyone,
Some months ago there were news about J2 filing lawsuits against companies
using fax-to-email technology, as they claimed it was their patent. They had
also won some cases, until someone filed a counter lawsuit against them
based some other grounds but again related to
T.38 is not a codec. A codec has one input and one output. T.38 is an
interactive protocol. This, however, has nothing to do with what I said.
If you use G.729 in the same asterisk as my spandsp library, you are
breaking my licence conditions.
Steve
Rob Hillis wrote:
T.38 is a codec in
Steve Underwood wrote:
I thought * was still not capable for T.38 gateway operation. Doesn't
beta 4 just added T.38 termination? And, I believe it misses out some
key elements of doing that properly. Note that T.38 termination is an
addon, so it can't be used with, say, G.729.
The only
Michelle Dupuis wrote:
Will the built-in T.38 support eliminate the need for spandsp? I'm
curious how this will affect iaxmodem.
Why on earth would you want to eliminiate spandsp? (which app_fax from
asterisk addons appears to use).
___
--
Steve Underwood wrote:
T.38 is not a codec. A codec has one input and one output. T.38 is an
interactive protocol. This, however, has nothing to do with what I said.
If you use G.729 in the same asterisk as my spandsp library, you are
breaking my licence conditions.
Steve
I should hope
I think you are missing something.
Steve means that since its in add-ons its probably a GPL addition and
not compatible with the g729 licensing.
A t.38 gateway involves more than origination and termination, those 2
are pretty easy and do not involve any modems, the gatewaying is the
harder
On Sat, 2008-02-23 at 10:09 +1100, Rob Hillis wrote:
So far I've never run into anything that's even close to the
speakerphone quality of the Polycoms. There's no comparison on the
speakerphone between the Linksys phones and the Polycoms - it's chalk
and cheese, but by the same token that
Wow, an answer phrased in the form of a flame...
A more supportive tone might actually encourage the Asterisk userbase to
grow!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thomas Kenyon
Sent: Saturday, February 23, 2008 8:22 AM
To: Asterisk
Zeeshan Zakaria wrote:
Hello everyone,
Some months ago there were news about J2 filing lawsuits against
companies using fax-to-email technology, as they claimed it was their
patent. They had also won some cases, until someone filed a counter
lawsuit against them based some other grounds
You can try using the asterisk -r -x CLI command
This allows you connect to the asterisk on the machine u run the command.
As for APIs have I have no ideas. May be the seniors can help you.
Thank you.
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On Sat, 2008-02-23 at 10:28 +0100, Fons van der Beek wrote:
I guess we are back to the fundamental problem: no asterisk generated
sounds on the external call
Do you have any T1/E1 cards in your system that aren't configured? If a
zaptel card isn't taking interrupts, that would cause this same
Jared YES
That seems to be the problem!
A very very long time ago I installed a X101P (an original one) and
forgot about it.
After issuing a modprobe ztdummy, indications on the outside line
indication work as they should.
After that i configured my X101P the way it should be
I used TelIAX for a while and was happy with the service. I used it
for testing before we connected to our PRI...
http://www.teliax.com
On Feb 23, 2008, at 7:22 AM, Zeeshan Zakaria wrote:
I posted the same question on asterisk-biz mailing list but didn't
have much response. So I am
I've had some serious issues with Teliax as of late with their new
Denver server. DTMF issues, IAX2 connection issues, and major latency
issues. They are blaming it on 1.2 vs 1.4 asterisk compatibility
issues. I have had zero problems with their old servers.
Voicepulse has been WAY
Michelle Dupuis wrote:
Wow, an answer phrased in the form of a flame...
A more supportive tone might actually encourage the Asterisk userbase to
grow!
Okay, if you really want a more constructive answer.
The addition to asterisk was an API change to allow app_fax from
asterisk-addons to
I'm using vitelity and junction for incoming, voipjet and voicepulse for
outgoing. You can set your outgoing caller-id with either provider; you
can not set your name. vitelity is the only provider I know of (other
than consumer-grade providers like vonage) who provide incoming CID
name.
I really like Junction Networks. They've been very good to deal with.
Michael
On Sat, 23 Feb 2008 11:30:35 -0600, Jay Milk wrote:
I'm using vitelity and junction for incoming, voipjet and voicepulse for
outgoing. You can set your outgoing caller-id with either provider; you
can not set your
--- Anthony Francis [EMAIL PROTECTED] wrote:
Have you tried placing the sip registrations in a db
using realtime?
I'm not that sure I want to use realtime because I
would then depend on the sql service never failing (I
could use clustered active-active MySQL but that
sounds overkill, or maybe
Pardon my ignorance but I understand that DUNDi
lookups (*CLI dundi lookup [EMAIL PROTECTED]) reveal if a given
extension is served by some host, ie. if it's
present in its dialplan. It does not say if it's
registered or not.
Is this correct?
On Sat, 2008-02-23 at 12:20 -0800, Vieri wrote:
Pardon my ignorance but I understand that DUNDi
lookups (*CLI dundi lookup [EMAIL PROTECTED]) reveal if a given
extension is served by some host, ie. if it's
present in its dialplan. It does not say if it's
registered or not.
Is this correct?
I must have started reading this thread after you reported that you
actually had an AUDIO problem rather than a RINGBACK problem.
The issue you experienced is a common one. Someday I hope Digium fixes
that bug/design flaw.
Fons van der Beek wrote:
Jared YES
That seems to be the problem!
I'm hoping someone can give me a little dialplan assistance. Here is my
scenario...
I currently have an ATT T1 connected to a Nortel Optn 11. I recently
purchased a Rhino system with a Rhino dual T1 card. What I want to do is
insert the Rhino box between the CO and the Nortel on the T1 so I can
Gleim, Jason wrote:
second T1 (Zap/g1). We have a number of DIDs that come in on that T1 and
I need them all transparently bridged for the time being.
[custom-nortel]
exten = _N.,1,Dial(Zap/g1/${EXTEN})
Anything that comes in will go right back out again.
Best regards,
Trevor Peirce
--
but your support was superior Eric!
tnx for your help!
Eric Wieling schreef:
I must have started reading this thread after you reported that you
actually had an AUDIO problem rather than a RINGBACK problem.
The issue you experienced is a common one. Someday I hope Digium fixes
that
Facing problem in installing asterisk-addonsThis was fixed in the latest
release of the add ons.
Subject: [asterisk-users] Facing problem in installing asterisk-addons
Hi,
I have installed GNU gatekeeper. Then I am trying to install asterisk addons.
I gave make and then make clean. I
About the only reason for eliminating SpanDSP is compatibility with the
GPL license. Remember that /any/ feature added to the free version of
Asterisk is going to be added to ABE as well - ergo, the licensing of
any libraries required need to be compatible with a /non/-open source
license.
T.38 is for all intents and purposes a codec. It's purpose is to
re-encode a fax transmission as a data stream to be re-assembled at the
other end as if it were a fax call. Seems to me to be pretty close to
the definition of a codec to me.
Your original comment was that you cannot use T.38 and
I've had some serious issues with Teliax as of late with their new
Denver server. DTMF issues, IAX2 connection issues, and major
latency issues. They are blaming it on 1.2 vs 1.4 asterisk compatibility
issues. I have had zero problems with their old servers.
Interesting... I've got several
If I'm using an Iax trunk between two sites what is the suggested number of
calls to pass over this trunk before I run in to problems?
Also is this number based on a per peer basis or all Iax calls going through
the server in general?
I know this will depend on the bandwidth I have between the
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