How do you want Switch to appear to Asterisk.
1. As an extension. Then use an FXS connection to a CO line input.
2. As a CO line. Then use an FXO connection to an Extension output.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTEC
Where do you get this new version of bristuff,I had a look on the
webpage and there's only RC3
On Wed, 2005-02-09 at 08:58, Peer Oliver Schmidt wrote:
> Altus Snyman wrote:
>
> > We have a quad bri card,installed on fedora core1,downloaded the latest
> > bri-stuff that download asterisk 1.0.3 and
Hi,
I have a sipura 2000 ATA connected to an asterisk server on the local
network, and the POTS line connected to asterisk using a X100P clone, when
calling remotely through the X100P (incoming call), the phone attached to
the sipura device always rings like it should, however sometimes it does
not
Michael Bielicki wrote:
>>conservation. (And yes, the latest to get is bristuff_0.0.2RC5 [RC6
>>seems to be for quadbri and octobri cards, only])
dIf you reread his email, he is stating that he has a quadbri
Oops, you are right. Note to self, don't write ML messages before first
coffee.
Note to O
You should wait till late afternoon whenb rc7 will come out
(hopefully). Also check if the new callerid behaviour in 1.0.5 fits
your needs
cheers
Michael
On Wed, 9 Feb 2005 08:23:59 +0100, Michael Bielicki <[EMAIL PROTECTED]> wrote:
> dIf you reread his email, he is stating that he has a quadbr
dIf you reread his email, he is stating that he has a quadbri
On Wed, 09 Feb 2005 07:58:38 +0100, Peer Oliver Schmidt
<[EMAIL PROTECTED]> wrote:
> Altus Snyman wrote:
>
> > We have a quad bri card,installed on fedora core1,downloaded the latest
> > bri-stuff that download asterisk 1.0.3 and zapte
Andrew Thompson wrote:
I am about to start a program that will be generaging sip device
configurations for sip.conf. My current sip.conf contains friend entries
for each SIP device connected to asterisk.
Should I even be attempting to split these in to seperate user/peer
devices?
Can two entr
Altus Snyman wrote:
We have a quad bri card,installed on fedora core1,downloaded the latest
bri-stuff that download asterisk 1.0.3 and zaptel 1.0.3 and libpri 1.0.3
All installed and working.BUT
after 5min+ of talking it just drops the calls?
Are you sure the call get dropped? We have a similar pro
Steven Critchfield wrote:
astfax allows you to create an email to fax gateway.
Are we going to see some integration of astfax with Courier-MTA/IMAP?
If you look at the instructions, you only need to make a some form of
default matching rule to catch all phone numbers and then pipe the
resulting mes
Good day all
What is the file sip_notify.conf for
Thanks
Altus
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> > Has anyone seen this message trying to install an TDM400.. spurious
> > 8259A interrupt: IRQ7
>
> I used to get this message a lot on my computer (before I even heard
> of Asterisk, mind you). When I looked into it and asked people
> questions, I was told that it is a harmless message.
I even
> Why?
>
> When you register with another provider, the use or not of MD5 Auth is
> up to him/her...
True, all I wanted was not to have plaintext passwords in sip.conf
(demonstrational purposes)
BRGS
Tomek
> > Hello Everyone!
> >
> > I just want to make sure if such a mess could work for sip
http://www.digium.com/index.php?menu=software_products
> -Original Message-
> From: Daniel Corbe [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, February 08, 2005 9:00 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] g729
>
> Can someone kindly point me to an RTFM in the f
Hi All,
I have a question about Fastagi because I can't get
it to work for some reason. Everytime I execute the
fastagi command, i get an error:
my extensions.conf:
..
exten => 1000,1,agi(agi://some_ip_address)
..
output from asterisk console:
-- Executing AGI("Zap/1-1",
"agi://some_ip_a
O did not have a look at it yet,I got the one from a week ago,how is
aterisk 1.0.5?
On Wed, 2005-02-09 at 08:04, Michael Bielicki wrote:
> hmmm the latest bristuff uses asterisk 1.0.5 so it can't be laast, can it ?
>
> cheers
>
> Michael
>
>
> On Wed, 09 Feb 2005 07:24:34 +0200, Altus Snyman
hmmm the latest bristuff uses asterisk 1.0.5 so it can't be laast, can it ?
cheers
Michael
On Wed, 09 Feb 2005 07:24:34 +0200, Altus Snyman <[EMAIL PROTECTED]> wrote:
> Good day all
> We have a quad bri card,installed on fedora core1,downloaded the latest
> bri-stuff that download asterisk 1.0.
Hi Mustafa N. Deeb,
Monday, February 7, 2005, 3:58:08 PM, Вы писали:
===8<==Original message text===
Mustafa N. Deeb> Hi
Mustafa N. Deeb> Has anyone got the 7902 phone work with asterisk , the only
thing I was able
Mustafa N. Deeb> to do with it, is to dial from it..
Good day all
We have a quad bri card,installed on fedora core1,downloaded the latest
bri-stuff that download asterisk 1.0.3 and zaptel 1.0.3 and libpri 1.0.3
All installed and working.BUT
after 5min+ of talking it just drops the calls?
Any reason why?
Please help
Thanks
Altus
_
I want to connect an asterisk box to a typical pbx
switch. What kind of interface i must use: FXS or FXO?And
why?
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No they are all on the same network,example server 192.168.0.250 and
phone(look at the warning) 192.168.0.250
On Tue, 2005-02-08 at 16:33, Kanuri, Seshu (Company IT) wrote:
> -Original Message-
> Good day all.I get the warning message on my system,this is for a snom
> 220,it repeats this
Matt,
I have the same issue and someone told me on IRC that broadvoice did not
allow the caller ID to be changed like you want. If you do get this to
work please share with the group so we can benefit.
Thanks!
Randy
Matt Schwartz wrote:
How do I get the incoming caller id to work correctly? I
Does Anyone on-list have any experience with these gateway devices?
>From their literature they claim to provide a mix of FXO/FXS SIP
interfaces. They also make specific claims about overcoming up to 20%
packet loss.
I'm always on the lookout for a better FXO.
Michael
--
Michael Graves
Hi All,
I was just reading through Info Week while on a flight and happened
upon an brief piece about a new VOIP security intiative worked up by a
handful of the usual suspects; Alcatel, SMU, NIST, Symantec, etc. All
of this begs the question of can't we get just do this as a user
community?
I un
On Tue, February 8, 2005 11:26 pm, Brian Dingman said:
> Was does your sip.conf look like for this Sipura?
Doh! I'll go slap myself now.
--
Paul A. DugasDugas Enterprises, LLC
[EMAIL PROTECTED]1711 Indian Ridge Drive
p:404-932-1355 f:770-516-4841 Woodstock, GA 301
When I try to load iax.conf I get (*-1.0.5):
loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/iax.conf:
cannot open shared object file: No such file or directory
Can anybody point out what is broken?
It was working OK few days ago but after recent Gentoo upgrade all of a
sudden it iax.co
Was does your sip.conf look like for this Sipura?
On Tue, 8 Feb 2005 22:56:15 -0500 (EST), Paul Dugas
<[EMAIL PROTECTED]> wrote:
> Running * CVS with a SPA-841 (0.9.5) and can't seem to get the message
> waiting light to come on automatically. There is a control in the web
> interface to turn it
Hi Manny
I have a sipura 3000 connected to asterisk and I must say, its not a bad
sollution. The units seem to be pre-configured to the US phone system and you
need to do some work to get them working properly, namely the hangup tone
detection...
The audio levels aren't too bad, default they cer
We found the trick was:
put the call on hold
Dial the number
Talk to other person
Press transfer when ready
Later,
PaulH
-Original Message-
From: Ulexus Silverthorn [mailto:[EMAIL PROTECTED]
Sent: Wednesday, 9 February 2005 1:12 PM
To: Asterisk Users Mailing List - Non-Commercial Dis
Running * CVS with a SPA-841 (0.9.5) and can't seem to get the message
waiting light to come on automatically. There is a control in the web
interface to turn it on and off (seems rather curious to me but whatever).
Has anybody got an idea as to where I may be going awry?
TIA,
Paul
--
Paul A.
OK, I've spent way more time than I wanted to on
getting
an x100p clone to work in Australia. I'm happy to
consider
other (more functional) options.
Does anyone have an opinion on both the Sipura
3000 and
other Digium cards (like the TDM400P)?
I need something that works with no much f
Here's a more complete call log:
-- Executing Goto("SIP/19544342000-b9c0",
"main-inbound|eglobalphone|1") in new stack
-- Goto (main-inbound,eglobalphone,1)
-- Executing SetVar("SIP/19544342000-b9c0",
"ALERT_INFO=Bellcore-dr2") in new stack
-- Executing SetAccount("SIP/19544342000-b9c
Hello,
I've inherited a (now) broken asterisk implementation. It seems as if
there are currently codec tanscoding issues in this box. Specifically
I am receving calls from a SIP proxy in G.729 and attempting to
transcode them to ULAW.
My asterisk installation was working up until yesterday.
Th
How do I get the incoming caller id to work correctly? I
have a broadvoice line going into my asterisk box. My dial plan then routes
the call to extension 1000. However, instead of the caller id from the
incoming call, I see the caller id number 1000 from the extension? How do I
correct
Tracy R Reed wrote:
On Tue, Dec 07, 2004 at 12:02:08PM +0100, Thorben G. Jensen spake thusly:
I cannot get the transfer button to work on a Snom 190, I cannot get the
# to work either.
I have a Snom 220 with a non-working transfer button. Not sure what the
problem is. Also need to figure out how
On February 8, 2005 07:53 pm, Steve Kann wrote:
> Glad it's working for you, Peter..
Seems to be working for me too; I'm using both 2532 and 3400. Your iax2 test
pktloss patch moved my build to /opt/asterisk/vCVS which caused me some
consternation but it's all good now. :-)
-A.
__
> Is there a way to either pass the date and time along from my POTS
> line going through a TD400P with one FXO and one FXS to my phone?
> Or even have * send the date and time through the caller id when
> that extension is called??
It should already pass the time from the Asterisk system. Mine
Peter Svensson wrote:
On Tue, 8 Feb 2005, Roy Sigurd Karlsbakk wrote:
how can I tune SIP jitter? is it possible today in asterisk?
I assume you are asking for how to alleviate the effects of jitter on
the
RTP audio streams initated by SIP? Asterisk currently only has a jitter
buffer fo
I have a regular telephone that when hooked to a standard POTS line, the
incoming callerid signal sets the time on the phone. I do this because
the phone has no internal battery and any little power blip causes the
time to reset to 1/1/98.
Is there a way to either pass the date and time along from
On 8 Feb 2005, at 22:54, Mike Dent wrote:
Phone numbers beginning with a '1'? Surely not, they should all start
with a 0 :)
Mike
It depends on the country!
At the end of the day, as long as the string is decipherable within the
data transcript within the switch, ending in an exgress route the
di
By the way, in case it matters, I don't have any
digium cards or SIP phones attached to the Asterisk
setup. Right now, it's all IAX, just receiving
incoming calls.
Cheers,
BeOnIce.
--- "Chamberland-Larose, Guillaume" <[EMAIL PROTECTED]>
wrote:
> If the asterisk process is hung up you should be
>
On Wed, 2005-02-09 at 00:37 +0100, Remco Barende wrote:
> Could you post your configs please? I need the same for my home setup :)
Sure, but there's nothing sophisticated:
extensions.conf
---
[myqueue-in]
exten => s,1,Queue(myqueue25)
exten => s,2,Macro(vm,${MYEXTEN})
exten => s
Guillaume,
The problem definitely is that the whole machine locks
up. Even at the console itself, I cannot get it to
accept typing! Even a Ctrl-C or Ctrl-Alt-Del or any
other such combination doesn't work.
I'll check the wiki for information about deadlocks,
thanks.
Cheers,
BeOnIce.
--- "Chamber
If the asterisk process is hung up you should be able to debug it. If
the whole machine is hung up this is a totally different issue isn't it.
If you're running linux and the machine locks up that often, you must
have a hardware problem. Or maybe you just think the machine is locked
up while it act
Looks really cool :)
My company requires that for every fax we send we get a printed status
report that includes the number we sent the fax to, the number the other
fax reported, time+date, tx time and if the fax was sent ok or not plus
(and here's the catch) a smaller image of the pages we faxe
I believe to do this effectively, I will have to modify asterisk's Dial
command to accept a timeout like 'L'. The difference would be that this
new command would allow asterisk to find this variable in shared memory
instead of locally. This would allow the AGI script as well as the
asterisk call
I am trying to set up a simple Asterisk server. All
it's going to do for now is to act as my voicemail
box. I've got a DID from Voicepulse, and am using IAX
(I'll get to SIP someday when I want to circumvent the
phone company for long-distance, but for now I'd be
happy to get a trial version of Ast
- Original Message -
From: "Stefan Gofferje" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, February 08, 2005 9:12 PM
Subject: Re: [Asterisk-Users] More complicated huntgroups / delayed ringing
Kevin P. Fleming schrieb:
Chris Wade wrote
Could you post your configs please? I need the same for my home setup :)
It's a bit silly for my friends to hear that they are nr. x in the queue,
annoying even, no friend wants to be put #2 :)
Thanks!!
On Wed, 9 Feb 2005, Bruno Hertz wrote:
On Tue, 2005-02-08 at 22:50 +0100, Stefan Gofferje wrot
We have a mix of Polycom IP600 and Sipura SPA2000 devices across our
network. I have noticed that the response times for the Polycom are
significantly higher than for other devices. I have also tested Cisco
and Snom hard phones. All of our phones are on T1 links back to the *
server.
My times
Thanks for the great howto, makes life a lot easier
Do you know if the florz patch also helps to prevent bristuff from
'dropping' the ISDN line? I have a problem that after a random amount of
time bristuff loses connectivity. At first incoming calls do not work
(they get information tone as if t
jbebeau wrote:
OK - I should know this... How does someone call in and pick up there
messages remotely?
Jon
Hint: Post a new thread properly, and those people who had stopped
reading this thread might see it.
Hint2: GIYF, asterisk check voicemail
--
Andrew Thompson
http://aktzero.com/
http://de
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Mike Nugent
>Sent: Tuesday, February 08, 2005 5:31 PM
>To: asterisk-users@lists.digium.com
>Subject: [Asterisk-Users] astcc with multiple access
>
>
>
>
>I'm looking at astcc and it seems that setting up a scr
Adrian Chapman wrote:
It's bizarre, isn't it?
Andrew sends mail to the list which sends to "Boris" whose mail server
returns it to the list manager, which interprets it as... a problem of
Andrew's - and sends Andrew his password automagically in the bounce so
anybody in the middle can frig aroun
On Tue, 2005-02-08 at 22:50 +0100, Stefan Gofferje wrote:
> Queues are not too bad but lacking an important feature. As far as I
> could figure out, they couldn't just ring without answering.
Hum? I'm running * 1.0.5, and Queue rings without prior answering the
line. Although most queue examples
[EMAIL PROTECTED] wrote:
I think you might be missing the point here. SER is a raw SIP processor.
So for a second throw everything you know about Asterisk + SIP out the
window and go back to vanilla SIP. Getting used to a B2BUA in the call
path kinda beats some of the raw power of SIP up. Think
On Tue, 2005-02-08 at 16:11 -0600, [EMAIL PROTECTED] wrote:
> Tim Burt wrote:
>
> >PLUS...
> >Numbering your extensions in the "100" to "119" range (or for larger
> >environments 1000 to 1199) will provide the cleanest interface. This is
> >because a leading 1 indicates a long distance call, and
On Tue, 2005-02-08 at 23:22 +0100, Peer Oliver Schmidt wrote:
> Ken Jones wrote:
> > astfax allows you to create an email to fax gateway.
>
> Are we going to see some integration of astfax with Courier-MTA/IMAP?
If you look at the instructions, you only need to make a some form of
default matchin
<< Oh, that's cool. I presume it will only assign the priorities
monotonically increasing from the last assigned priority? Are there docs
anywhere on this? I checked the archives and voip-info.org... I guess the
changelog in CVS might have it, eh?>>
Yes, it really is cool. "n" does increase p
I am about to start a program that will be generaging sip device
configurations for sip.conf. My current sip.conf contains friend entries
for each SIP device connected to asterisk.
Should I even be attempting to split these in to seperate user/peer devices?
Is there(should there be) a convention
Phone numbers beginning with a '1'? Surely not, they should all start
with a 0 :)
Mike
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I have several Polycom IP-500’s and a few of the Cisco 7960’s connected
to an Asterisk test box. When I add qualify=yes to the sip.conf and
then enter “sip show peers” on the console I get, on average, 85 ms for
the Polycom phones while the Cisco phones are half that. This is on a
LAN. Acr
Peter Svensson wrote:
The other queue implementation, icd, already has exactly this in the
agent_priority_group distributor.
Peter
Looks like ICD has made some progress recently then, I'll need to check
it out again.
[as he heads off to the world of cvs...]
-chris
__
Can I have multiple classes like with mpg123?
yes..
slayer => mp3:/var/lib/asterisk/mohmp3/slayer,-z
tool => mp3:/var/lib/asterisk/mohmp3/tool,-z
Doesn't work:
Feb 6 01:03:06 WARNING[8815]: res_musiconhold.c:354 moh0_exec: Unable
to start music on hold (class '') on channel SIP/6004-1416
Someon
I'm looking at astcc and it seems that setting up a script that will
allow multiple people to access a calling card simultaneously would be
fairly difficult. Before I endevour to develop this, has anyone already
done it/looked at this/can point me in the right direction?
Thanks.
--
Mike Nugen
Here's a thought. A user is dialing a local number (local for them)
but accidently dials it 1+.. Is it LD?
No, quite simply each of the prefix digits in the local exchange, would
route back to the same routing case, which would be setup with the same
charging record, effectively nulling the ad
Matthew Boehm wrote:
With all of these caveats, it seems to me that a SER->Asterisk solution
isn't that great. If anyone else out there can show me otherwise...
Thanks,
Matthew
I think you might be missing the point here. SER is a raw SIP processor.
So for a second throw everything you know abo
Michael Manousos schrieb:
asterisk-oh323-0.7.0 is for Asterisk CVS.
How did you manage to compile it with Asterisk-1.0.3?
Hi,
sorry, I just checked again: I'm using asterisk-oh323-0.6.4
on that machine, where asterisk-stable runs.
Use Asterisk-1.0.3 with asterisk-oh323-0.6.5.
Ok, I will try it ri
My PBX seems to have just started showing wierd codec negotiation problems.
I'm not all of a sudden getting this on certain phone numbers on my system:
Feb 8 22:19:19 NOTICE[1125329728]: channel.c:1683
ast_set_read_format: Unable to find a path from ULAW to G729A
Feb 8 22:19:19 NOTICE[112532972
On Tue, 8 Feb 2005, Kevin P. Fleming wrote:
> I believe this is correct, the call must be answered before app_queue
> can handle it. However, how many customers do you think would sit there
> for 3 or 4 minutes of ringing with no announcement messages or anything?
> I doubt very many would last
Ken Jones wrote:
astfax allows you to create an email to fax gateway.
Are we going to see some integration of astfax with Courier-MTA/IMAP?
--
Best regards
Peer Oliver Schmidt
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Dear all,
The new version of DIAX (0.9.10a) is ready to be downloaded.
The web site and the help file are updated too.
What's new comparing with 0.9.9g:
- independent codec configuration for each registration server;
- use control chars in the dial string to automatically send some DTMF codes
aft
Hello Florian,
I now experience a lot of drop outs during a conversation. They last 5
seconds and more, but eventually the sound comes back (if the other side
has not hang up).
Can you tell us wether you are using ISDN or VoIP phones, and if you are
using VoIP phones, how the network is configure
On Tue, 8 Feb 2005, Roy Sigurd Karlsbakk wrote:
> >> how can I tune SIP jitter? is it possible today in asterisk?
> >
> > I assume you are asking for how to alleviate the effects of jitter on
> > the
> > RTP audio streams initated by SIP? Asterisk currently only has a jitter
> > buffer for IAX, n
On Tue, 8 Feb 2005, Chris Wade wrote:
> Kevin P. Fleming wrote:
> > I've been considering doing this as well... something like a "dial
> > list", with a delay before dialing and a timeout for each entry.
>
> Might also help to implement this type of 'strategy' for the 'Queue'?
> Just another lit
On Tue, 8 Feb 2005, Chris Wade wrote:
> Exactly, but if you look, * queues don't quite measure up to these
> requirements. The asterisk-ICD project is getting there, but still too
> immature I think.
After a bit of man-handling icd works really well. We are doing weird and
wonderful things wi
Tim Burt wrote:
PLUS...
Numbering your extensions in the "100" to "119" range (or for larger
environments 1000 to 1199) will provide the cleanest interface. This is
because a leading 1 indicates a long distance call, and the number
following a leading 1 cannot be a "0" or a "1" for long distance.
Steven Critchfield wrote:
it was
someone elses mail server failing, and bouncing the message to Digium.
But some how or another, Mailman is attributing the bounce to the
original poster though.
It's bizarre, isn't it?
Andrew sends mail to the list which sends to "Boris" whose mail server
returns i
Chris Wade wrote:
Or something like that. It would even be great if the app could - given
the previous flow - detect that SIP/101 was busy on the first try and
immediately start trying SIP/102, skipping the delay. Did you follow
what I meant by that?
Yes, I understood your description, it was
Stefan Gofferje wrote:
So, if I ran a 0900 support-no. for people without contracts, they
probably won't like paying EUR 3,60 per minute to hear music and - after
three or four minutes - be told, there's still no agent free to answer
the call...
I believe this is correct, the call must be answe
Stefan Gofferje wrote:
Hum... sounds pretty much like you needed an ACD rather than a simple
immediate / delayed calling huntgroup...
Regards,
Stefan
Exactly, but if you look, * queues don't quite measure up to these
requirements. The asterisk-ICD project is getting there, but still too
imma
> Here's a completely different and rather ambitious idea...
> What about replacing (or complement) the Dial and Queue apps by a
> meta-ACD app. Something like a scipting language. Something REALLY scary
> flexible and powerful. Something where another wiz can write a GUI
> for... So, with this,
This happens when you put the wrong value in extension.conf
in my test ;line belpw ypu see I have @broadvoice after the number
exten => 708,2,Dial(SIP/[EMAIL PROTECTED],15,m)
you probably have @sip.broadvoice.com
the @broadvoice needs to be in your sip.conf
ie
[broadvoice]
Randy
Puddle wrote:
I'
On Tue, Feb 08, 2005 at 04:03:50PM +0200, Doug Reid - Stormcorp wrote:
> Hi
>
> Try going into "vi /etc/profile" insert the lines in brackets.
>
>
>
> USER="`id -un`"
> LOGNAME=$USER
Generally LOGNAME is set by login, sshd
how to make g.729 preferred, but failover
to gsm
I've purchased a few g.729 licences, and would like to set up iax.conf
such that g.729 is used if they are available, but then it fails over to
gsm.
I'm not sure how to specify such a preference. I'll let the server transcode
from ulaw (from th
On Tue, 2005-02-08 at 14:48 -0600, Marco Castillo wrote:
> I recently have purchased a new TE110P card, that provides a single T1/E1
> port. I have installed it and everything works fine, except for the dial
> tones. When I made a call from a SIP phone to a channel in the TE110P, I
> receive no dia
Kevin P. Fleming wrote:
Chris Wade wrote:
Might also help to implement this type of 'strategy' for the 'Queue'?
Just another little idea, I still like the previously discussed
options as well.
Possibly, although it could be done with a queue by just directing the
queue to call a Local channel th
Kristian Kielhofner wrote:
Andrew,
1) - When you signed up you were given the option for a monthly
password reminder. That is what you recieved.
This is a probe message. You can ignore this message.
The Asterisk-Users mailing list has received a number of bounces from
you, indicating that th
OK - I should know this... How does someone call in and pick up there
messages remotely?
Jon
- Original Message -
From: "Kristian Kielhofner" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, February 08, 2005 3:20 PM
Subject: Re: [Asteris
Chris Wade wrote:
Might also help to implement this type of 'strategy' for the 'Queue'?
Just another little idea, I still like the previously discussed options
as well.
Possibly, although it could be done with a queue by just directing the
queue to call a Local channel that then distributes the c
I recently have purchased a new TE110P card, that provides a single T1/E1
port. I have installed it and everything works fine, except for the dial
tones. When I made a call from a SIP phone to a channel in the TE110P, I
receive no dial tone. When I receive a call in a SIP phone from a channel in
th
Kevin P. Fleming wrote:
Yes, DialList() was my work-in-progress name already :-) It might be
possible to merge RetryDial()'s functionality in, but I haven't really
looked yet. I wasn't planning on working on this seriously for another
couple of weeks...
Follow me and see if you understand why I
dorian logan wrote:
Hi,
Regarding these high bandwidth CODECs - is it possible to upgrade
asterisk to record at a higher quality bit rate too
Yes, with development.
- is Asterisk based on a 8Khz system.
Yes, presently.
We would like to stream calls from SIP phones to the internet at a
higher qu
Hi There,
I've made a small Asterisk + ISDN HFC_PCI + Music-on-hold + Soft fax HOWTO.
I think it's pretty useful for people that are getting started with
Asterisk. If you're interested check http://www.bartroos.com/asterisk/.
If you have any questions or suggestions for this HOWTO, feel free to
On Tue, 2005-02-08 at 14:20 -0600, Kristian Kielhofner wrote:
> Andrew Thompson wrote:
>
> > Twice in the last week or so, I've received a message similar to the
> > attached.
> >
> > A portion of the attachment that's attached is not in English. Is this
> > my mail server failing, or someones
Kevin P. Fleming wrote:
I've been considering doing this as well... something like a "dial
list", with a delay before dialing and a timeout for each entry.
Might also help to implement this type of 'strategy' for the 'Queue'?
Just another little idea, I still like the previously discussed options
Chris Wade wrote:
Delay before dialing and timeout for each entry? I think I follow your
choice in words there, your saying - forgive possible stupid syntax -
something like 'tech/id[delay:timeout]' where delay determines how long
before dial even tries that device and timeout is the same thing
Andrew Thompson wrote:
Twice in the last week or so, I've received a message similar to the
attached.
A portion of the attachment that's attached is not in English. Is this
my mail server failing, or someones who's on the list?
Andrew,
1) - When you signed up you were given the option for a mon
On Tue, 2005-02-08 at 14:06 -0500, Andrew Thompson wrote:
> Twice in the last week or so, I've received a message similar to the
> attached.
>
> A portion of the attachment that's attached is not in English. Is this
> my mail server failing, or someones who's on the list?
>
Did you read it in
Stefan Gofferje wrote:
Andrew Thompson schrieb:
I'll be playing around with Local/ some in the next few days(now that
I more understand what it's for).
I had a thought about your problem. Given the dialplan above, have you
tried adding a Wait(15) to extension [EMAIL PROTECTED], so it doesn't
st
I am not sure I got you Amine. But if the GSM gateway is receiving the
call. The callerid depends on the GSM Gateway settings and if it can
deliver it. (I know that analog ones do not deliver CallerID )
Please elaborate
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
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