On 25 March 2010 02:42, James Lamanna jlama...@gmail.com wrote:
Hi,
I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of
D-Channels going down and then coming back up (See below).
Read all the discussion about many spans - and I've run 16 E1 spans in one
box, and run 8
Problem is that the port 80 you are talking about is a TCP port. Voip
(iax and rtp) use UDP
On 2/11/10, mosbah.abdelkader mosbah.abdelka...@gmail.com wrote:
Thank you Jamie for your good reply.
It is a very good idea to hava the media and control transported over the
same port with IAX
On 9 February 2010 06:42, Muro, Sam resea...@businesstz.com wrote:
Hi Team
Can someone advice me on how i can lower the load average on my asterisk
server?
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.10.1
asterisk-1.4.25.1
2 X TE412P Digium cards on ISDN PRI
Im using the
2010/1/12 Jeff LaCoursiere j...@jeff.net
That is so not true. FreePBX has hooks in a million places to do custom
dialplan stuff - I do it all the time. I also link in custom AGI/AMI
applications, custom provisioning, custom LCR, and am even working with
one customer that has mastered making
2010/1/15 Peter Childs pchi...@bcs.org
Elastix, Trixbox, or AsterixNow, or DIY (ie Ubuntu or whatever
installed with OpenPBX, Asterix etc by hand)
I've got a new server to run Asterix on and want to get it working
quickly and yet be configurable in the future with out having to
reisntall
2010/1/15 Doug Lytle supp...@drdos.info
Decide if you are going to be a zealot for your preferred approach
That's a little harsh, wouldn't you say? Do whatever your most
comfortable with. But, to call me and those like me a zealot, for
offering advice that was asked for is a little off,
What you are missing is the new state-interface parameter to AddQueueMember.
You can't use functions in a hint exten.
Steve
On 12/14/09, Lenz Emilitri lenz.lo...@gmail.com wrote:
Hello all,
I am trying to set up a dynamic channel to be used as an Agent dialer for a
queue - you know, trying
Hi,
I need to build a simple, command-line method to generate a legal and
perfect RTP stream across a network link, and analyse it on the other side
and measure network performance. Want to do this for a number of links and
over long periods. I'm trying to characterise performance of various
2009/9/9 Armin Schindler ar...@melware.de
No, I didn't miss that. See my text.
I mentioned this because I think this might be the reason of the problem
and
the incorrect handling in jitterbuffer, if it is the jitterbuffer. It is
just a guess, since everything else seems to work good.
2009/9/8 James Mutuku listmut...@gmail.com
I have 2 sites. One(Site 1) has an asterisk PBx and the Other(site 2) has 2
remote soft phones. The latency btw both sites is btw 500ms-700ms. I know
this is a shot in the dark...but are there ways of improving the voice
quality for the remote
2009/9/9 Karl Fife karlf...@gmail.com
...of course you need one of these to dial SIP URI's or navigate IVR's from
the rotary mechanism.
http://www.oldphoneworks.com/rotatone-pulse-to-tone-converter.html
On Asterisk I don't think that's true. At least for IVRs on the local
Asterisk box,
In any event, the real problem is probably that you forgot to 'include
= parkedcalls' in your dialplan.
Steve
On 9/2/09, Lyle Giese l...@lcrcomputer.net wrote:
And now that the whole world of Asterisk has your sip user ids and
passwords, you should change all of the passwords that are in that
2009/8/30 jonas kellens jonas.kell...@telenet.be
I am totally not understanding this :
My IAX.conf :
register = BOX-YOCAN:pas...@remote_asterisk_ip yocan9...@89.31.97.186
On remote Asterisk :
*CLI [Aug 30 20:37:07] -- Registered IAX2 'BOX-YOCAN' (AUTHENTICATED)
at ip:4569
So this
2009/7/21 Jim Dickenson dicken...@cfmc.com
How can the first step of the extension be a playback when I do a verbose?
Because you have a exten = line that matches *9901 in the empl context or
some context that is included into the empl context before (above)
dorecord.
Steve
Hii,
Looking at this, your problem appears to be that you are diskbound. Note
the 60% wait time.
Use hdparm -t to find out the throughput of each of your disks. If its
not 20MB/sec or more then you need to look into the drivers you are using
for your disk. If you do have good disk throughput
Hi,
We have a customer who used a strong quad-core Xeon box to convert up
to 800 simultneous calls from SIP to IAX and trunk them to another
box.
So your requirement doesn't look like a big problem.
Steve
On 3/24/09, Christian Victor christ...@victormedia.de wrote:
Hi!
A customer of mine
Hi,
Are you sure that Verizon amswers the call? They should play that
message as 'early media' without answering, after which they cam clear
the call with an appropriate cause code.
That would work for you and still give callers the audible ,essage they want.
Steve
On 3/20/09, drew einhorn
Hi,
Xorcom make what you are looking for.
Steve
On 3/15/09, Duncan Turnbull dun...@e-simple.co.nz wrote:
Hi All
I am looking at a replacement for a hotel PBX which requires at least 60
analogue extensions.
I tend to use Sangoma equipment but haven't tried this many analogue
extensions
If there is NAT between the phone and * then that can be responsible.
Also, Eyebeam (et al)'s ICE setting causes this.
Steve
On 3/13/09, Mike Diehl mdi...@diehlnet.com wrote:
Hi all,
I've got a problem where many times, there is silence at the first 1-2
seconds of a call. Then it clears up
Hi,
I know i doesn't make practical difference, but often it is the far
end that is atually buggy, not out end.
A lot of the work in spandsp to increase success rate is to do with
workarounds for issues in the remote machine,
Steve
On 3/13/09, Marshall Henderson marshall...@gmail.com wrote:
To get busy state for a sip channel in 1.4 it appears the peer/friend
must have a call-limit.
Steve
On 3/9/09, Cary Fitch ca...@usawide.net wrote:
Running an earlier version of Asterisk (1.2), we were using Hints to show
busy extensions on other (SNOM) phones.
When we went to version 1.4
It can only be acoustic echo. Asterisk doesn't cancel that - it's the
phone's job.
Maybe it will fix it to reduce volume of the phones.
Steve
On 2/27/09, Bruce Komito bru...@bagel.com wrote:
I know the subject of echo has been discussed ad nauseum, but I think I
have a somewhat unusual
Hi,
As others have mentioned, the 'n' is a pattern char.
I have a system that uses similar tricks to yours. What I did about
this issue was to change the pattern match chars to be upper case
only. Drop me a line if you want the patch.
Regards,
Steve
On 2/12/09, Chris Bagnall
Everybody is talking about other products. But yes, the Xorcom will
handle all ports active, supports a high density connector at the
back, looks just like standard Zap/Dahdi ports to Asterisk, rack
mounts nicely and much less $$ than the other solutions.
Steve
On 2/10/09, Erick Perez
2008/10/23 Kristian Kielhofner [EMAIL PROTECTED]
Most of the anything but simple PAT devices I've seen that implement
any SIP specific fixups usually end up breaking something along the
line. Unless the product is from a company where SIP is their core
competency (like Ingate, or /maybe/
2008/10/23 Udo Schacht-Wiegand [EMAIL PROTECTED]
For a door opener on an Astribank FXS port we need a loop current of 24.5mA
.
It does not function with the Astribank now, the dialtone becomes quiet
immediately after pressing the button on that device.
I've seen a limit of 23mA in the zaptel
2008/9/12 Justin Coffi [EMAIL PROTECTED]
Does your box run on the Mr. Fusion power supply?
My box is plugged into the national power network. Oh right - I'm in South
Africa and the national power supply company is Eskom. I see your point
that it couldn't have been up 38 years.
For
xx-montague-gardens*CLI show uptime
System uptime: 38 years, 37 weeks, 4 days, 10 hours, 47 minutes, 11
seconds
Amazing. Especially considering:
[EMAIL PROTECTED]:/var/log uptime
09:58:14 up 18:42, load average: 0.21, 0.09, 0.02
Steve
___
--
2008/9/12 randulo [EMAIL PROTECTED]
On Fri, Sep 12, 2008 at 11:13 AM, Tim Panton [EMAIL PROTECTED] wrote:
I'd guess the battery on your motherboard has died so it is going back
to 1970 at
boottime.
Why do hide the truth, Tim? It's much more likely the motherboard
traveled back 38 years
2008/9/12 Doug Lytle [EMAIL PROTECTED]
Stephen Davies wrote:
Why don't you guys believe that my Asterisk has just been up for 38
years?
Because Mark was born in 1977 and he's 31.
Oh dear. Maybe this will help: ;-) :-)
Steve
2008/7/6 Grey Man [EMAIL PROTECTED]:
From what I can gather the suggestion from the FS approach is that
each Asterisk channel should be handled after by it's own unique
thread and save the need for any locking on the channel data
structures in the first place.
After a quick grep, there are
I have a network of offices using Asterisk that are connected via IAX2
trunks. The trunks work great for a day or two then for no reason at all one
end of the trunk will become UNREACHABLE while the other end is still
connected. The oving nly way to fix the problem is to shutdown Asterisk
I'm sure that an Asterisk developer can chime in and give several examples
of how Asterisk uses its threads to increase scalability. That said, there
will be a point where the number of core/CPU's won't be the bottleneck so
adding more won't help anything.
Asterisk is highly multi-threaded
On 20/01/2008, Michael J. Liberatore [EMAIL PROTECTED]
wrote:
They are extremely upset because calls are being randomly bridged for no
rhyme or reason. They say that callers will call in and sometimes get
connected with other callers, or they will be in the queue and then be
talking to
On 21/01/2008, Steve Davies [EMAIL PROTECTED] wrote:
b) Attended. Wait for the call to answer, Press transfer, you will be
ordered a list of calls in the order 3, 2, 1 - This is 100% wrong. The
call you want is LAST in the list. If you have no CID, or have
forgotten the CID of the caller, you
Hi Steve,
Please look at my asterisk-dev post from a few minutes ago about
dcontext and dst where the behaviour changed in a bad way in svn trunk
recently.
Thanks,
Steve
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
On 09/06/07, Deepak Naidu [EMAIL PROTECTED] wrote:
Ya, I have done that, below is zapata.conf. Also we had an TMP card with
analog lines. SIP cals were great on them. now when we switched over.
SIP calls have echo.. which shouldnt be at all.
If you are getting echo on pure SIP to SIP calls,
On 08/06/07, Asterisk [EMAIL PROTECTED] wrote:
Would a good 1 gBit switch be enough to handle that (Asterisk box would
be connected to that switch with 1 gBit connection, and computers with
Microsoft RTC Library would be connected with a 100 mBit connection)?
Alex: 30 concurrent calls will be
Hi Matthew:
Your environment sounds quite challenging and I'd be interested in the
analysis of what is limiting the throughput.
I agree that there's no easy way to distribute and single queue across
multiple boxes.
But here is a scaling idea for you. We've used it successfully to
handle a
On 01/06/07, Douglas Garstang [EMAIL PROTECTED] wrote:
I previously worked for a company that did some heavy load testing with
Asterisk on multiple core Sun systems. We saw that no matter how many
cores you threw at Asterisk, it always used ONE core to process calls,
even at very high loads.
On 01/06/07, Matthew J. Roth [EMAIL PROTECTED] wrote:
Mon Apr 2 12:15:01 EDT 2007
Idle (sar -P ALL 60 14) (60 seconds 14 slices)
Linux 2.6.12-1.1376_FC3smp (4core.imminc.com) 04/02/07
12:24:01 CPU %user %nice %system %iowait %idle
12:25:02
Hi,
I want to quickly mention that I've had great success with running
Asterisk in the under-appreciated Linux-VServer environment.
This is not so much a virtualisation environment as a system partioner
on steroids. Nothing to do with running windows on Linux and
suchlike, but a good way to
On 14/05/07, Salvatore Giudice
[EMAIL PROTECTED] wrote:
Try switching to a Sangoma card. You won't have anymore IRQ issues once you
abandon Digium hardware.
Its not true, by the way.
I've assisted more than one person using a Sangoma who was having
issues caused by interrupt stuff.
And it
On 29/04/07, Noah Miller [EMAIL PROTECTED] wrote:
I've heard of a device that acts as a failover for a PRI line so you
can plug a PRI into two different devices and have the PRI failover if
one device fails. Unfortunately nothing like this is commercially
available today.
Sounds like
On 09/01/07, Nigel Kendrick [EMAIL PROTECTED] wrote:
I've had a play with a Nokia E70 - the 'bar' version of the E61 and gave up!
Menu navigation is dire - I went through hoops trying to get SIP working - I
know from others it can be done, but I bailed out when I realised that to
put these
On 05/11/06, James Harper [EMAIL PROTECTED] wrote:
Even in this configuration, with my impedance settings set to the
Australian standard of 220+820||120nf, and the PSTN and PAP2 echo
cancellers enabled (or not, and all combinations of) I get local echo as
soon as I pick up the handset (I hear my
On 26/10/06, Guillermo Salas M. [EMAIL PROTECTED] wrote:
What about the bandwidth used for both protocols? Is IAX using less or
more bandwidth than SIP?
I'll give you an actual measured result.
A trunked IAX2 link, carrying 30 simultaneous calls using
variable-bit-rate Speex - we saw 7
On 02/11/06, Matthew Mackes (Webmail) [EMAIL PROTECTED] wrote:
As far as Snow- They look very cool, and I love almost everything Linux
based- PDA's, PVR,s, everything- but, I wonder if it will need to be
rebooted every once in a while to stay happy- Every phone that is SIP
has an OS- so, its
On 15/06/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Who said I was a C programmer?
Speaking for myself, I just assumed that you understood that the
behaviour of an open-source application was the result of contributed
code. Your message read to me like something of a demand that
someone
Hi,
I have a strange problem on a single customer's PRI. He can't call
certain destinations, receiving an incompatible destination ISDN
cause code back from the network.
I'm sure that the PRI is misconfigured by the telco; but they (as
always) insist there is nothing wrong. Another Asterisk
On Thu, 27 May 2004, Vladyslav wrote:
Good day All.
Is there a way to pass DTMF signals to AGI script during conversation ?
Actually here what I want to make:
Users are usually dial using dialplan and when someone press *4 (during
conversation) I want to have agi script to deal with
On Tue, 25 May 2004, jo wrote:
Sorry, no solution but same problem. Downgrading brings this message on
Suse9.0, 2.4.21:
[app_txtcidname.so]May 25 23:28:42 WARNING[16384]: loader.c:240
ast_load_resource: /usr/lib/asterisk/modules/app_txtcidname.so:
undefined symbol: ast_get_txt
May
On Tue, 25 May 2004, Dan Cunningham wrote:
Like some others on the list spandsp is segfaulting asterisk when recieving
a fax. I'm on debian testing/unstable with freshly checked out asterisk
CVS and sandsp. My libtiff version is 3.6.1.
You need an older libtiff - v3.5.7.
Steve
On Tue, 25 May 2004, Terry Goodwin wrote:
Thanks for offering to help with this.
I checked out the procedures and attempted this again without success.
Here is the end of the screen output when the compile fails.
gcc -02 -g -Include -I ../include -c -o app_rxfax.o app_rxfax.c
On Mon, 24 May 2004, usedcanon wrote:
I have a requirement for a setup with prepaid call credits.
I am aware of the two applications available (been researching for the past
week), app_prepaid and app_rateengine. However neither of the two sound like
exactly what I want. However I was
On Mon, 24 May 2004, Chad Brown wrote:
1.The 2 SIP phones can call MeetMe and have a conference but
cannot call each other. (Yes, they connect but no audio either
direction)
2.I have verify=yes in the sip.conf for both phones. Both phones
constantly go Unreachable. (However, the
On Sun, 23 May 2004, gARetH baBB wrote:
On Sun, 23 May 2004, Karl Dyson wrote:
Of course, although my wife is happy with the Cisco 7905s that have
sprung up around the house, she still likes the cordless DECT units we
have, and so they're plugged into an ATA186. Problem is, they no
On Wed, 21 Apr 2004, Jeremy Jones wrote:
I've been searching on an error I'm getting trying to compile against
uClibc, related to enum support. I found reference in an earlier thread
(http://lists.digium.com/pipermail/asterisk-users/2003-June/014176.html)
to a patch adding an Makefile
On Wed, 21 Apr 2004, Tom wrote:
It doesn't look very hard. FreeBSD supports recursive mutexes. It is
just a matter of getting the appropriate defines. I'm going to look at
this.
On my Gentoo system I had to add #define _GNU_SOURCE to lock.h just
before it #includes pthread.h.
That
On Tue, 13 Apr 2004, Alex Brett wrote:
Has anybody got any experience using an X100P on an NTL phone line in
the UK (I'm in an ex Cable Wireless area if that makes any difference).
The problem I'm having (and judging by the number of references to it
I've found searching it is a common
On Tue, 13 Apr 2004, Vic Cross wrote:
On Tue, 13 Apr 2004, Stephen Davies wrote:
I did some work a while back to add detection of the UK busy/hangup
signal on the line, but I never got it working well enough to depend
on it. The problem is that it is a single frequency tone. (The US
Hi,
I've installed spandsp library an RxFAX app on my Gentoo based *
server - here's a little report on the process.
Firstly - my Gentoo box had libtiff 3.6.0 installed, but did not have
/usr/include/tif_dir.h and tiffiop.h.
First thing I did was to copy the two missing headers from
On Sat, 20 Mar 2004, Zac Amsler wrote:
I know this issue has been address before, but I can not find someone who
has the answer.
I am trying to get my * server to authenticate directly to packet8.
I was very close to them actually giving me the information and possibly
using them for my
On Fri, 12 Mar 2004, Brian Capouch wrote:
I too am running 6 cards in my system, although not in a high traffic
capacity load environment.
So far my (limited) high-load simulations have shown no problems.
So - is it apocryphal that the Digium cards (drivers) won't share
interrupts?
If
On Wed, 10 Mar 2004, Fran Boon wrote:
Patch failed - this is what this output is showing.
As Matt said the patch needs modifying to patch cleanly against the
current version of the code...
You didn't read his mail properly.
Steve
___
Hi,
I thought it would be neat to put my SIP/IAX reachable systems into
the ENUM system.
But reading about it I see that its rather centrally controlled within
the ITU.
My country code (+27) is not delegated. My country has a monopoly
telco whose only interest in VOIP is to keep it all to
On Tue, 17 Feb 2004, Rich Adamson wrote:
I have an full-time Internet connection with a limited amount of traffic per
month included in the subscription.
What can I do to reduce the registration traffic with IAXTEL which makes
about 10MB/h?
There is any way to keep the registration
On Tue, 10 Feb 2004, Alex Lopez wrote:
[outsidedialtone]
exten = s,1,Playtones(350+440) ; US standard dialtone from indications.conf
exten = _X,1,SetVar(FIRSTNUM=${EXTEN}) ; Had to get the first digit dialed
and hold on to it!!
exten = _X,2,StopPlaytones()
exten =
On Thu, 5 Feb 2004 [EMAIL PROTECTED] wrote:
Hi
I wonder if anyone has a fix or any advice for the IAX2
jitter buffer.
My internet connection here in South Africa has an
international ping time of 550ms +- 50 ms. According to the
scientific approach I would like to add a 100ms jitter
On Thu, 5 Feb 2004, Chris Lee wrote:
On the subject of South Africa
What are the laws regarding using the Internet to carry telephone traffic?
What are the laws regarding connecting digium kit to Telkom equipment?
As I recall they are quite restrictive, have they been eased up a bit?
The
On Fri, 30 Jan 2004, Dustin Goodwin wrote:
Actually I believe this is one of the few things that can be done
without worrying about the state(s) PUC coming down on your head. Since
your users are in another country the state PUC cannot consider you
providing a telephone service in their
On Fri, 30 Jan 2004, Steve Rodgers wrote:
Oops! I forgot the leading underscore. Use this version below.
Steve.
exten =_ NXX,1,Dial(Zap/1/$EXTEN)
exten = _1NXXNXX,1,Dial(Zap/1/$EXTEN)
And reaching us wot is in the rest of the world...? ;-)
Steve
On Thu, 29 Jan 2004, Dmitry Mishchenko wrote:
All example of installing Asterisk shows running it under root user.
Why is that? Can it be run under regular non-privileged user account.
Sure - with the right permission tweaking.
I made a group telephony.
Had to fiddle with permissons and
On Sun, 25 Jan 2004, Christopher Lee wrote:
I've had a closer listen to 400*17 through the handpiece rather than just on
speaker phone, and I get the feeling that the Australian ringing tone must
have been tweaked slightly, perhaps with the introduction of the newer
Ericsson AXE exchanges?
On Sun, 25 Jan 2004, Christopher Lee wrote:
The original indications has 400+17/400, but I find that sounds more like
two beeps (which could possibly be confused with the Australian
congestion/busy tones).
Shouldn't it be 400*17?
Steve
___
On Wed, 14 Jan 2004, Steve Underwood wrote:
That must have been an FSK modem. Most advanced modems completely loose
sync on the first sample slip. The sample slip causes a jump in phase,
and phase is critical to the correct operation of most modems.
It was V.22. No error correction or
On Wed, 14 Jan 2004, TC wrote:
What are the practical effects with in-correct clock sync
-like to you hear odd buzzing, or dropped voice or gaps of audio ??
Old-fart anecdote about this - in the early 80s we had some 1200bps
modems that we used to connect to client sites. When our phone
On Sun, 11 Jan 2004, John Todd wrote:
zed
Thanks!
knots per hour
Pretty sure the measure is just knots. IE 40 knot wind, or the wind
will be 40 knots.
Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Wed, 7 Jan 2004, calvis wrote:
I am trying to calculate bandwidth needs. Is 1 T1 Line able to provide
488.5 Gigabytes of traffic for 1 month based on a 30 day week? I did my
calculation as follows:
1.544 mps * number of seconds in a minute(60) * number of minutes in a hour
(60) *
On Wed, 7 Jan 2004, Sjur Eivind Usken wrote:
We have setup an asterisk box to let everybody call into the university
internal network, but I get unexpected hangups when doing an outbound call
from SIP to the ISDN interface, and it happens from 20 seconds to some minutes into
the
call.
On Wed, 7 Jan 2004, Tilghman Lesher wrote:
On Wednesday 07 January 2004 06:06, Dawid Mielnik wrote:
Hello,
I can not seem to be able to get StripMSD and Prefix to work for me
in extensions.conf. This is an example of what I have:
exten = _050.,1,StripMSD,1
exten =
On Wed, 24 Dec 2003, denon wrote:
I've been having issues getting FWD to work. I posted this same Q to the
FWD forum (no responses yet), but I was hoping someone here had some insight:
My setup is like this:
sip.conf:
register = 21542:[EMAIL PROTECTED]/6002 ; Free World Dialup
On Wed, 24 Dec 2003, Iain Stevenson wrote:
I have exactly this problem and posted a bug report to FWD about a week ago
- no response yet. It's bizarre that FWD recognises you to dial another
user but not to call outside their network. Sounds more like a FWD problem
than a * problem to
On Sun, 21 Dec 2003, Darren Nickerson wrote:
In the case of a physically-disconnected ZAP extension, the Dial application
succeeds, moving on to the next step in the dialplan. That is much more in
line with my expectation.
With an X100P card, diconnecting the card from the line results in
On Sat, 19 Jul 2003, Darren Poulson wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I've got my developers kit from telappliant and got a machine up and running
to become the house phone system. Most things are working now, such as
incoming calls, call transfer, call
On Thu, 26 Jun 2003, Dan wrote:
There is nobody with an X100P in Europe having this issue related to the
PSTN Caller ID?
Please help!
Well - my X100P doesn't pick up Callerid from my UK line. But I
always assumed that it was just not compatible with UK-style Callerid.
Steve
On Thu, 19 Jun 2003, Holger von Ameln wrote:
Hi,
Stephen Davis offered to send me a patch that leaves out enum support.
That would at least solve the undefined references to res_ninit,
res_nsearch and res_nclose in enum.c.
Cheers,
Holger
Hi,
Here it is, attached. Adds a setting
On Wed, 18 Jun 2003, Bradley Greep wrote:
Hello Tielman Koekemoer
E1 is used in the world except for North America and one or two other places.
It consists of 30 speech or data channels and 2 signalling (1 for framed signalling,
and one for channel signalling)
E1 is superior to the
Hi,
I'm having trouble using Ringing with a SIP client. I'm trying to
give the caller the impression that the line hasn't been answered,
whilst listening for various extensions to be dialled.
Here's is the extension:
exten = s,1,Wait(1)
exten = s,2,Answer
exten = s,3,DigitTimeout,3
exten =
1. 8.2.6.1 Sending a Provisional Response says that UASs SHOULD NOT issue
a provisional response to non-INVITE requests.
From my message yesterday * appears to be sending a SIP/2.0 100 Trying to
X-Lite's REGISTER request before sending the SIP/2.0 200 OK message.
Is this correct?
Why not have dial just dial, then have applications like WaitForAnswer,
WaitForDisconnect etc...?
This would give more granularity to the call flow control and allow
someone to get brave and write a WaitForHuman or whatever.
Hmm... I can't think of too many instances where the
Hi,
Has anyone done anything with the Linux advanced routing stuff to give
SIP and IAX traffic priority?
What I have in mind is a high-pri queue for voip traffic, all the rest
in another queue that gives way to the VOIP stuff.
Thanks,
Steve
___
On 10 Jun 2003, Emanuele Pucciarelli wrote:
Il mar, 2003-06-10 alle 17:19, Stephen Davies ha scritto:
Has anyone done anything with the Linux advanced routing stuff to give
SIP and IAX traffic priority?
What I have in mind is a high-pri queue for voip traffic, all the rest
On 10 Jun 2003, Emanuele Pucciarelli wrote:
That is not entirely correct. There is an output queue, and pfifo_fast
is the default (see the LARTC Howto, 9.2.1.1). But you are right when
you say you need something to slow down the data;the simplest choice
should be addingthe Token Bucket
On Sat, 7 Jun 2003, Daryl Jones wrote:
I experienced the exact same symptoms but didn't have the confidence
to post my experience to this list because of my lack of experience with
Asterisk. I restored the June 1 version from CVS and the problem went away.
There's definitely a problem in
On Sun, 8 Jun 2003, Brian Capouch wrote:
FYI to anyone else who may be experiencing random hangups; I removed the
busydetect=yes lines from the conf files on my asterisk servers, and
haven't had a hangup since.
I had done that once before and it didn't seem to have much of an
effect,
On Sat, 7 Jun 2003, shido wrote:
This is the sip debug when the call went through
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Content-Length: 157
Content-Type: application/sdp
CSeq: 1 INVITE
From: sip:[EMAIL
Hi,
Does anyone know anything about the callerid format that NTL uses here
in Cambridge, UK. This is the former Cambridge Cable, who is
sometimes different from the rest of NTL.
They did say that only some equipment works with their switch. I
hoped that they might use US-style CID, which would
On Mon, 2 Jun 2003, Mark Thompson wrote:
I can use an ata186 to connected directly to n2p through
sip.net2phone.com without any special settings.
I can connect from * to iconnecthere, but, whatever I try from * to n2p
produces SIP/2.0 401 Unauthorized
(Can forward the full * sip log and
On Mon, 2 Jun 2003, Tilghman Lesher wrote:
First, they're going to have to be MMX. The 133 might be, but the 75
is definitely not. Normally you don't want to go much below a 200MHz
processor for a base-level system; you could certainly try something
slower, but not without MMX
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