Richard J. Sears wrote:
Hey Everyone -
I am having a problem that is keeping me awake at night.ok, so maybe
not keeping me awake, but it is frustrating. :-)
I am running Asterisk 1.0.7 on Gentoo (2.6.10-gentoo-r6) on an Intel
700Mhz box with 512MB of RAM.
The system is very light, with maybe
John Goerzen wrote:
Hi,
I recently encountered an odd situation: the network cable to my
SPA-841 got unplugged while it was in the midst of a call. I got it
re-plugged in about 30 seconds, and the phone rebooted. The phone
showed no evidence of the previous call in progress and worked like
normal
Bernie wrote:
can that number be reduced? I'm looking at a system that would be
deployed to remote offices over fairly limited bandwidth links and need
to find a way of balancing quality vs. bandwidth constraints.
Yes. Read up on the various codecs and how much bandwidth they use.
_
Actually about 80k-82k when you take into account UDP and RTP overhead
and assume you are using SIP. Single IAX2 call may be a little less.
multiple IAX2 calls using trunking will be a lot less.
In fact, this question is answered on
http://www.digium.com/index.php?menu=documentation
specifical
Randy Paries wrote:
Thanks for the info
OK my first questions
I have edited my zaptel.conf
fxsks=1-2
loadzone = us
defaultzone=us
I have two X100P cards installed
When I run /sbin/ztcfg
ZT_CHANCONFIG failed on channel 2: Invalid argument (22)
Did you forget that
Steve Mann wrote:
From what I have read, you made a small mistake, if you are not using Digium
hardware, but want to use MeetMe of Music on Hold, you still require a
timing source, regardless of kernel.
A Zaptel Timer has not been required for MoH for at least a year.
__
Glenn Powers wrote:
I keep getting this error every five minutes:
Apr 4 13:35:00 NOTICE[20551]: res_musiconhold.c:309 monmp3thread:
Request to schedule in the past?!?!
Apr 4 13:35:01 NOTICE[20551]: res_musiconhold.c:309 monmp3thread:
Request to schedule in the past?!?!
Apr 4 13:35:01 NOTICE[20
Maik Hassel wrote:
Hello everybody,
I am having trouble setting up a SIP/analog phone gateway. The SIP
phones are working, just the Zaptel card doesn't seem to work. I am
using the zaptel TDM400P with one FXO module on the last bank (should be
channel 4 I suppose).
When I try to dial out (either
Tobias Jönsson wrote:
On Thu, 31 Mar 2005, Peter Svensson wrote:
It would not be very hard to add both features to libpri. Libpri
already has a function to decode and dump the time/date information.
If I remember correctly the time/date IE should be added to the SETUP
messages. I have been think
Rod Bacon wrote:
This is quite interesting.
I tested calls to 2 mobiles that I knew were off, and not diverted to
voicemail. 1 with Telstra, the other with vodafone (I'm in Australia).
Via ISDN, both calls were shown as unanswered by asterisk. When the
calls went to voicemail, the call was deeme
Mark Halverson wrote:
exten => _1NXXNXX,1,SetGroup(${CALLERIDNUM})
Try using ${ACCOUNTCODE} and make sure the account code is unique to
each phone.
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On Apr 3, 2005 8:56 PM, Ian Hailey <[EMAIL PROTECTED]> wrote:
Hello all,
I was hoping to be able to call a mobile and if it is un-reachable for
whatever reason (e.g. switched off) then I was expecting an unobtainable
response that would be detected in Asterisk. It seems that the operator
(Virgin i
Courtney Couch wrote:
If I were to buy 20 did's how do I know within asterisk which number was
dialed? (like say I want a few of the did's to ring specific extensions
if they are dialed and others to go through the menu)
Is there any ${var} that has the number dialed in on? (that would be
optim
Brandon Patterson wrote:
Level 3 does DTMF inband DTMF. Period.
If he's using IAX he's not talking directly to Level 3.
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Brian Litzinger wrote:
On Fri, Apr 01, 2005 at 12:12:57PM -0600, Eric Wieling aka ManxPower wrote:
Brian Litzinger wrote:
> iax.conf:
[general]
bandwidth=high
allow=all
jitterbuffer=no
tos=low
register => 1234567:[EMAIL PROTECTED]
[livevoip]
type=friend
secret=1234567890
deny=0.0.0.0/0
Brian Litzinger wrote:
> iax.conf:
[general]
bandwidth=high
allow=all
jitterbuffer=no
tos=low
register => 1234567:[EMAIL PROTECTED]
[livevoip]
type=friend
secret=1234567890
deny=0.0.0.0/0.0.0.0
permit=217.160.244.186/255.255.255.0
context=from-livevoip
sip.conf:
I have dtmfmode=inband for both sip
Kib Eki wrote:
Thanks, problem solved, I found somethind in this mailing list! Wrong
extensions.conf entry.
extensions.conf:
exten => 0237482,1,Dial,CAPI/@301:0237482,5,tr
?? But, what does ",5,tr" mean ??
"5" tells Asterisk to hang up if the call is not answered in 5 seconds.
"t" tells Asterisk
Eric Wieling aka ManxPower wrote:
Does anyone know what I need to do to use patlooptest? I have what I
think is a T-1 loopback plug in the card (1-port, TE110P), but I still
see a red alarm. Is this normal? I don't even know where to start for
this.
From Digium Support:
You will ne
[EMAIL PROTECTED] wrote:
I don't believe that Digium is as narrow minded as suggested by the poster below.
I don't believe Digium is interested in using a web based forum just
because a few people don't understand how to manage e-mail.
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Does anyone know what I need to do to use patlooptest? I have what I
think is a T-1 loopback plug in the card (1-port, TE110P), but I still
see a red alarm. Is this normal? I don't even know where to start for
this.
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A
present != ship
Zoa wrote:
Yes they do, it was presented at von. Its a little daughterboard for
te4xxp cards.
Jerry wrote:
Digium has a hardware echo can?
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Scott Bussinger wrote:
forums). If only we could get people to quit posting in HTML email, life
would be grand. :)
Mozilla has an option to view ALL messages as text. I use that. I
suppose I should not. People that post in HMTL should not get my help.
Maybe I can use procmail to send an autom
Jerry wrote:
On Mar 31, 2005, at 8:01 AM, Zoa wrote:
cpu load on te4xxp cards is very low, and now that they have echo
cancellers as add-ons cards, it will be even lower.
I can't speak on hardware compatibility as i never tried a sangoma card.
(But i can say that in the last year i've never had an
Brett, Gary wrote:
can receive calls, but the problem comes when making calls, when I dial out,
it gets to the British Telecom exchange and brings back the BT message "The
number you have dialled has not been recoginised" . so the first thing I did
was to make sure that the number was the full tele
Garrett Nelson wrote:
Ok, I am still working on getting this PolyCom phone working with Asterisk.
I have been looking all over, but I have not been able to find the username
and password for the web interface on this phone.
I found some site that said it was Polycom and spip, but that does not work
Anton Krall wrote:
Any problems with RTP or voice just on one side?
So as long as you use some STUN server, the RTP packets have the right IP.
Did you install your own stund or are you using a public one?
You didn't have to use SER at all right?
Setting nat=yes does pretty much the same as a STUN s
Noah Silverman wrote:
How??
There is a nice big "hold" button on the phone. How do I re-configure
the IP500 so that * handles the hold???
Upgrade your Asterisk first. A MoH bug was recently fixed. (which you
would have known if you were following the mailing lists)
the IP500's hold button wor
Joseph wrote:
On Sun, 2005-03-27 at 18:22 +0200, Torsten Krueger wrote:
Hello,
On Sun, 27 Mar 2005, Jim Sturtevant wrote:
What are the advantages of the Digium PCI cards for FXS ports vs standalone
ATAs?
You can _reliably_ fax with them.
That maybe it.
Though with new addition: NVFaxDetect featur
Bill Kervaski wrote:
I've tried two completely different scenerios.
1) Debian (sarge) with the Asterisk 1.0.5 package.
2) Redhat 9 with Asterisk CVS 1.0.7+
I can't get the AGI "STREAM FILE" command to work with a simple bash
script. I can get other AGI commands to work like "SAY NUMBER 123" etc.
Bart Van Daal wrote:
That's because someone suggested it earlier on the list. So
I installed the ztdummy driver.
Could you please tell me what is needed to playback sound files?
Nothing. It Just Works.
You call into your Asterisk server, dial the extension for the Playback
or Background, or whate
Bart Van Daal wrote:
Thanks for your answer,
I've compiled and loaded 'ztdummy' but still no sound.
here's the relevant portion of lsmod:
ztdummy 2464 0 (unused)
wcusb 19552 0 (unused)
zaptel178752 0 [ztdummy wcusb]
i810_audio 28
Justin Ramsey wrote:
Does anyone know what a sample extensions.conf and/or sip.conf would
look like to place a call on mute? Also, what does a sample
extensions.conf file look like for three way calling.
That is all handled by your SIP phone. Check the docs for your SIP phone.
__
Parker, Blake (MIS) wrote:
Can Asterisks properly handle outbound Enhanced 911?
Can the Ford F150 handle blue?
Neither of the above question makes any sense without additional
information.
Asterisk supports one of the 6 or so ways a PBX can support E911.
If you provide the details of what specifi
Tom wrote:
This is what I have suspected all along is that the signaling and timing
constraints on the PRI are such that you basically need asterisk running as a
real-time process. The whole point of the thread (in my mind) is if there is
anyway to cause X to not run as such a real-time process so
Sys Admin wrote:
I am setting up a new asterisk based call center. I just read:
http://www.voip-info.org/wiki-IAX+versus+SIP
After reading this and other google results for "IAX vs SIP" is there
any reason why i should use SIP anywhere !!
Because most equipment doesn't support IAX
--
Always do righ
Alessandra Grasso wrote:
My objective is to estimate the performances of *
How much the trancoded can influence the performances?
Thanks,
show translation recalc 30
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
___
A
It means the caller hung up in the middle of the voicemail app.
Anton Krall wrote:
So far, nobody has been able to tell us what this error means.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber
Sent: Lunes, 21 de Marzo de 2005 02:54 a.m.
To: Ast
Brian McCrary wrote:
Hello,
I'm wondering if anyone has benn able to successfully get g726-16
passthrouhg to work? I am wanting to use this codec instead of g729 as
I'm running out of DSPs using a high complexity codec on the Ciscos. I
would think it would work just as g729 does, which has been w
Kevin P. Fleming wrote:
Matt Gibson wrote:
This is what I'm sending from my dhcpd server.
option ntp-servers 10.x.x.x;
option tftp-server-name "ftp.x.x.x";
option time-offset -18000;
Keep in mind that using TFTP for a Polycom boot server is sub-optimal,
because you have to renam
Tyler wrote:
I think you're looking for the 'ChanSpy' application that seems to have
inexplicably vanished from the asterisk CVS..
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20ChanSpy
If anyone has any info on this, let me know as I'm in a similar
situation.
As far as I know ChanSp
cmisip wrote:
No path to translate from SIP/fwdpulvercom-dd5a(2) to Phone/phone0(1)
I don't know why the above message is printing codec numnbers, rather
than names. *shrug*
"show codecs" will tell you what codec number are what codec name.
It appears that your Phone/phone0 is using G723.1. Look
C F wrote:
Now consider this (this works with the cisco 7960, even if you put a
7914 with it, it will still use all 20+ plus buttons this way, if CW
is disabled on the phone):
exten => 123,1,Dial(SIP/${EXTEN},30,tr)
exten => 123,2,Voicemail(u${EXTEN})
exten => 123,3,Playback(goodbye)
exten => 123,4
Asterisk is not a SIP proxy.
Wei Su wrote:
We encouter a situation where we need to use SIP info to convey infomation
for one end point to another endpoint. I use asterisk to do the test and
find asterisk does not forward the SIP info to another endpoint, but act as
UAS and returns a 4xx error mess
Anton Krall wrote:
What do you think?
CPU0
0: 16148159 XT-PIC timer
1: 4 XT-PIC keyboard
2: 0 XT-PIC cascade
5: 0 XT-PIC usb-uhci
8: 1 XT-PIC rtc
10: 161351663 XT-PIC usb-uhci, w
Thomas Andrews wrote:
If I comment out the following line in zapata.conf I would expect
asterisk to "forget" the cli information for that channel when I reload:
callerid="Uniden Dead" <(256) 428-6125>
... but it doesn't; I have to restart asterisk for it to take effect.
The funny thing is that the
John Goerzen wrote:
Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
extensions.conf lines:
exten => s,1,SetVar(SET_EMERG_FLAG=0)
exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten => s,n,SetGlobalVar(EMERGENCY=1)
exten => s,n,SetVar(SET_EMERG_FLAG=1)
exten => s,n(di
Jason Becker wrote:
Daniel Burget wrote:
I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines
connected via TE405P. Everything works great, except MOH. I added an
exten with MusicOnHold(30), and it plays just fine. Conferences have
music when no one is in. I have SIP phones. When I
Richard J. Sears wrote:
Hey Everyone,
I am using NuFone for 866 inbound service and I am trying to figure out
the callerid part of it. Any call into my * system just shows "Toll Free
Call" and will not give me the calling party's caller ID info.
Is this just something I have to live with using NuFO
Once you run Dial from an AGI script, you lose control of the call via
the AGI script.
Jean-Michel Hiver wrote:
(obviously if you do other magic in your dialplan this needs to be
adjusted. The important part is the 'g' flag to Dial (go on after
hangup), and the NoOp which echos the dialstatus
David Zanetti wrote:
I've been beating my head a bit against the 1.0.6 Debian builds of
Asterisk, using an E100P (E1, single span) board.
In machines I've built in the past (back in 1.0.0 days), config I'm
using and that card and 1.0.0 driver combo worked fine.
ztcfg reports no problems:
SP
Neil A. Hillard wrote:
Using X-Lite to dial extension 400, I hear it ring and then get answered
and I hear about 0.1 of a second of the on hold music and then silence.
If I use the 'line 1' button to put the call on hold and then take it
off again I hear another 0.1 of a second of the music. This
Kanuri, Seshu (Company IT) wrote:
Thanks for the pointers. Here is my Features.conf where I have tried my
best to use Asterisk to give away control. I have enabled ## as the
combination key for Asterisk (in quick succession) to retain control,
but otherwise ignore the key presses.
I don't run CVS-H
Kanuri, Seshu (Company IT) wrote:
I am using Asterisk 1.06 Stable.
When I dial my Mobile Number to check Voice Mail or my Bank Account
Phone Access Number, the IVR System on the other end asks me to enter
*2378 to transfer to an attendant.
But When I press *2378, Asterisk tells me that it cannot tr
[EMAIL PROTECTED] wrote:
Hello,
I upgraded my office from Asterisk 1.0.0 to Asterisk
CVS-HEAD-03/13/05-13:14:04 this weekend, and are now
experiencing some problems accessing voicemail. The web based interface
works fine, in addition to dialing 8500,
which is mapped to:
exten => 8500,1,Voicemail
César Davi Ávila do Nascimento wrote:
Talk about skype is forbidden, but to be impolite is allowed...
Great list!
Skype does not interface with Asterisk in any way whatsoever. You
could just as well have asked if someone knows what RNA sequence 42 in
the turnip genome is for. About as many peop
Ron Joffe wrote:
Hey folks
I have a new setup with a TDM400P for a pair of analog extensions and a few
SIP phones. We seem to be experiencing a bunch of "Crackeling" when talking
between the analog and SIP extensions.
Any ideas?
Yes. Check the suggestions given to the other guy that posted this
Raoul Bönisch wrote:
* Eric Wieling <[EMAIL PROTECTED]> [2005-03-14 16:56]:
Raoul Bönisch wrote:
Flash is an analog thing. It does not even apply to ISDN.
So how does the "R" key on my ISDN-telephone work then?
I suspect it sends an ISDN specific "put call on hold&quo
Roman Zhovtulya wrote:
Hello,
I wonder if I would have to sacrifice anything if I set "NAT=yes" for
all sip clients I have, regardless of whether they are behind the NAT or
not.
The idea is to have the setting that works regardless of whether the
user is behind the NAT or not, since I'm not sure wh
Brian McCrary wrote:
Hello,
I'm trying to run an ATA behind a NAT device, and am confused on exactly
what the qualify config option does, other than send NOTIFY packets.
Outbound calls work fine, but inbound calls do not go through. With
qualify=yes and nat=yes, my show sip peers looks like:
77
Colin Anderson wrote:
I'm going to do a deployment of LocustWorld MeshBoxes in some of our remote
locations. Build 90 comes with Asterisk 1.0, and our plan is to use the
MeshBoxes as a WAP for non-Asterisk uses but also to add a 2nd NIC to deploy
Snom's in the remote location. This works fine (was
Raoul Bönisch wrote:
Hello!
I'd like to Flash() a modem line (BRI) with Asterisk. It is a
passive ISDN-card connected to a hardware PBX. I use ISDN4Linux.
I recognised that unfortunately the Flash() application flashes
Zap devices only. Now I am wondering how I could flash Modem/ttyI0.
The source c
Robert Hajime Lanning wrote:
Robert Hajime Lanning wrote:
um, backwards. E-Mail to SMS. I have not seen the other way
around.
Both Cingular and Verizon supports both.
I have not tried this, nor have I seen any documentation mentioning
it. Do you or anyone else have a pointer for the info?
Espe
Robert Hajime Lanning wrote:
Well, as far as I know there is no such service in the USA. Take in
mind that SMS is not so popular in the states, email is, and every
cell phone in the US that I have seen that supports SMS, supports SMS
to email from the phone as well.
um, backwards. E-Mail to SMS.
Peter Svensson wrote:
On Sun, 13 Mar 2005, Robert Hajime Lanning wrote:
There are SMS sending gateways out there, but they are sending
only, no way to receive. This is fixed in the IM solution by
giving the "system" an account of its own.
Whatever gave you that idea? Most operators have an inter
On March 13, 2005 09:57 am, Nigel Burgess wrote:
[door]
exten => s,1,Dial (SIP31,15)
exten => s,2,Playtones(dtmf)
However the call hangsup before trying to play the DTMF tone.
When a Dial happens, the dialplan stops until the call is
disconnected. See "show application dial" to see how you can se
David Uzzell wrote:
I have seen the list of codecs for the ATA 186's but not sure if it was
100% or not.
I want to know really is it possible to run GSM or ilbc on them or is a
G729 lic the only way to get a low bandwidth codec?
This is the list of codecs that I have seen.
RxCodec and TxCodec—C
I have no idea. I live in the USA so I don't normally need busydetect.
Anton Krall wrote:
Why does busydetect actually drop calls while stile talking?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Viernes, 11 de Marzo de 2005
[EMAIL PROTECTED] wrote:
I upgraded a DVG-1120M to a DVG-1120S. Everything works great, but I'm
having some caller ID issues on incoming calls sent to the SIP device.
Using debug on the device, the caller ID looks fine - just as I set it in
Asterisk. However, the phone is showing "CID TRANSMISSI
Anton Krall wrote:
Guys, this is weird.. Today I started having some problems with calls been
dropped. Im suing X100p cards (clones) and I have this setting on my zatala
fle:
[channels]
[snip]
busydetect=yes
busycount=4
Can the echotraining be messing things? Do I need to enable callprogress or
so
Wiley Siler wrote:
Yep. I have run "sip show peers" and things look good.
I am pretty sure I could see all my registered phones yesterday via "sip
show registry".
But then again maybe I am spacing it. I am on AAH BTW and info in AMP
and in direct check of the confs checks out fine.
"sip show regi
Nenad Radosavljevic wrote:
Same problem here: if call come over ISDN PRI and it is for a SIP phone
that equals to strong echo situation, at the SIP end. Interestingly this
doesn't happen on all calls but it does on 95% of them. Asterisk load at
that moment is insignificant - 1 to 2 calls.
I hav
Wiley Siler wrote:
Hello all,
For some reason I am not showing registration in SIP.
Can anyone give me an idea what can cause this?
asterisk1*CLI> sip show registry
HostUsername Refresh State
You don't have any register => lines in sip.conf.
Maybe you are looking
Deti Fliegl wrote:
Hi there,
all that started by investigating what happens if SIP clients are
calling anonymously.
The problem: Every client who is registered as a regular user with
username and secret can fake any callerid in subsequent INVITEs.
Asterisk does not apply an accountcode or caller
James Murray wrote:
Can anyone offer any suggestions for quality hardware sip phones under
$150. Preferable one with a 2 line caller id screen and the ability to
disable call waiting. It would also be very useful if it had a good
voice echo cancellation built into the phone.
SIPura SPA-841
--
Calin Serbanescu wrote:
hello list,
last night i upgraded my asterisk box from 1.0.5 to 1.0.6 and my music
on hold did not work anymore.
my setup is ISDN (wct1xxp)->SIP (Audiocodes mp124) and reverse.
the system refuses to activate music on hold resource... i returned to
1.0.5 and it works fine ag
Ronald Wiplinger wrote:
Can anybody help me and explain me the following area codes:
1-340 US-USVI
1-670 US-CNMI
1-710 US-Governement
1-787 US-Poerto Rico
1-802 ~ 1-808 ???
1-939 US-Poerto Rico
1-600 Canada
Are the above codes are USA Continental tarrif (NuFone / Broadvoice ...
Puerto
Jim Gottlieb wrote:
I see that app_qcall has been replaced. We rely on this for some of
our applications. What has it been replaced by?
It was nice to be able to just dump files into
/var/spool/asterisk/qcall and have the calls be placed automatically.
/var/spool/ourgoing is where you toss you
Time Bandit wrote:
For SIP incoming/outgoing you normally need ports 5060 and the port
range 1-2 open.
At least it works in my setup.
Could anyone correct it if it's not exactly all the truth?
Just a little note that all these ports are for UDP, no TCP ports are used.
Also SIP devices are f
James Pooton wrote:
I think you're in luck :) Sipura has a new firmware out for the 841 now,
version 3.1.1a. (Quite the jump from 0.9.5) It includes -6db, 0db, +6db
options for input gain on the handset, headset and speakerphone now. Release
notes and downloads are available on their site.
The
Scott Bussinger wrote:
We're setting up some Sipura 841 phones and they're working pretty well, but
the microphone volume on the headset (not the handset) is too loud with our
Plantronics headsets. Is there some way to turn down the amplification on
the headset mic?
The microphones are picking up t
Ronald Wiplinger wrote:
Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
I need to setup all area codes for billing, but how can I do that easy
for North America and Canada, where I have only one price anyway.
Country code 1, and just exclude any Carribean nations that you need to
handle dif
Wilson Pickett wrote:
My problem is as well described on
http://www.voip-info.org/wiki-Asterisk+i+extension
Yup. I believe the 'i' reacts to dialed digits input during that
particular extension. There aren't any. That wiki page concludes that
you need a "fallthrough" extension like
_.,1,Playback(I
Eric_Doiron wrote:
Try specifying the contexts ... just an idea.
exten => 1,4,VoiceMail([EMAIL PROTECTED]&[EMAIL PROTECTED])
Remember these are voicemail.conf contexts, not extensions.conf contexts.
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Kristian Kielhofner wrote:
Eric Wieling wrote:
Kristian Kielhofner wrote:
Goto(my-internal-sipphones,1234)
Goto(my-internal-sipphones,8005551212)
The first argument is the context, the next argument is the
number. Your internal sip phones should be able to reach each other
via "exte
Kristian Kielhofner wrote:
Goto(my-internal-sipphones,1234)
Goto(my-internal-sipphones,8005551212)
The first argument is the context, the next argument is the number.
Your internal sip phones should be able to reach each other via
"extension" dialing and (hopefully) be able to reach outside n
Mike Dent wrote:
Makes you wonder how many *really* reliable VoIP providers there are out there?
Who would you trust to handle all your incoming/outgoing business calls?
None.
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Sip.conf
[ext1]
Context=phones
Mailbox=201
Voicemail.conf
[home]
201,password,name,[EMAIL PROTECTED]
[EMAIL PROTECTED]
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On Friday 04 March 2005 11:43 am, Nigel Taylor wrote:
> The line is fine. If I connect an analog phone I get ringtone and can call
> out. If I leave the phone connected and connect another wire from the
> socket into the X100P, I immediately lose the ringtone. I've checked The
> voltages and when t
On Friday 04 March 2005 02:00 am, Robert Rozman wrote:
> > exten => 42,1,SetVar(SAVED_EXTEN=${EXTEN})
> > exten => 42,2,Goto(marvin,27,1)
>
> thanks for help. I'd just like to be sure what happens if there is more
> than one concurrent calls. Is variable set up for each of them or is
> necessary t
On Thursday 03 March 2005 04:11 pm, Robert Rozman wrote:
> Hi,
>
> I'm trying to implement dynamic routing of incoming calls to local
> extension if previous outgoing call was unanswered.
> But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to
> 's-NOANSWER'. I guess this is normal, but I
Anton Krall wrote:
When you dialout using zap lines and sip phones, the sip connects to the zap
channel and then dials the number, on the logs its shows sip => zap channel
and when zap picks it up shows as answered but how can you really tell if
the dialed number was answered or busy?
If you are u
Julius Kidubuka wrote:
When I do click on the phpconfig.php link from
http://ip-of-machine/phpconfig/, it returns a page with the actual
contents of that file (phpconfig.php) and doesn't load the page. See
some
of the output below;
Try a simple php-script in this directory.
Something like this, nam
Robert Rozman wrote:
Hi,
I've updated my Asterisk 3 times with :
cvs checkout -r v1-0 zaptel asterisk asterisk-addons
and then do
cd asterisk
make clean && make && make install
make samples
make progdocs
and then when I run Asterisk I get :
Asterisk CVS-v1-0-02/11/05-01:46:25, Copyright (C) 1999-20
Kamran Ahmad wrote:
hi
If i remove "_." from my dialplan(extensions.conf).
application is invoked only once. otherwise
application is invoked again and again. any one know
what is the problem and how to make (global) dialplan
for all user agents.
When a call hangs up Asterisk will loog for an 'h' e
Philipp von Klitzing wrote:
Hi Eric!
and do NOT use bandwidth=
Why is that? I am curious...
Because bandwidth= just enables specific codecs. The specific codecs
enabled depend on the bandwidth= setting.
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C F wrote:
When there is no timing source then MOH will not work, however
asterisk can use the incoming RTP stream as the timing source, if the
device you are trying to use don't send packets when there is no
sound, then there is no RTP stream, therefore no timing source.
If Asterisk does not have
Marty Mastera wrote:
Hello:
I have read that music on hold requires a timing source (which I never
had to worry about previously since the server had zaptel hardware in
it)...now I'm configuring a server in a colo which has no zaptel
hardware.
MoH has not required a timing source for at least a y
Jerry Geis wrote:
I am running into a problem where I have a menu and I want the
user to enter # when they are done. However doing so then asks
to transfer. How do I disable that.
Don't use options on the Dial option that you do not understand.
Remove the "t" or "T" option from your Dial line.
___
Michael George wrote:
I see that 1.0.5 is out. I thought that if I am tracking cvs v1.0.x I would
always get the newest releases. However, I just did a fresh update and
install from cvs stable and it reports as only being v1.0.3.
Should I just be using the tarballs rather than the cvs -r 1_0? Or
Scott Bussinger wrote:
Has anyone found an administration manual for the Sipura SPA-841 phones? I
found a quick start guide and a user manual at the website (good thing
because there was _nothing_ in the box), but I haven't found a manual to
explain the more complex features in the web setup pages.
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