This really looks like we are missing a lot of the associated code.
PaulH
On Wed, 2008-02-20 at 00:28 -0800, Shaun R. wrote:
A call comes in and goes into the queue, the queue dials a sip channel using
a macro. The macro plays a set of options to the callee and if the callee
presses 3 it
:07 +1100, Paul Hales wrote:
I am guessing that 'not yet handled' is not good news.
PaulH
P[ 2] I IND :SETUP oad:383208100 dad:93409098 pid:415 state:none
P[ 2] I SEND:PROCEEDING oad:0383208100 dad:93409098 pid:415
-- Executing Answer(mISDN/2-1, ) in new stack
P[ 2] * ANSWER:
P[ 2
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Paul Hales
Sent: Tuesday, February 19, 2008 8:42 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] ISDN2 facility code...
I have just been given the answer -
exten = *44,1,Answer
exten = *44,n,Noop(Check Callforward
On Mon, 2008-02-18 at 08:11 -0500, Jared Smith wrote:
On Mon, 2008-02-18 at 16:35 +1100, Paul Hales wrote:
I spoke to a telco tech and he said I had to send a facility
codehuh?
Anyone with any ideas on this one?
I know there's a setting in zapata.conf called facilityenable -- have
state:CONNECTED
P[ 2] I IND :FACILITY oad:0383208100 dad:93409098 pid:415
state:CONNECTED
P[ 2] -- not yet handled
On Tue, 2008-02-19 at 10:45 +1100, Paul Hales wrote:
On Mon, 2008-02-18 at 08:11 -0500, Jared Smith wrote:
On Mon, 2008-02-18 at 16:35 +1100, Paul Hales wrote:
I spoke to a telco tech
I am trying to send 'codes' over an isdn2 link - such as *#24# - to
activate call forwarding.
But it doesn't work. I have tried sending it as a straight dial, and
also as a DTMF string...but no luck...
I spoke to a telco tech and he said I had to send a facility
codehuh?
Anyone
We upgrade 2 of our Aastra 57iCT to the latest firmware (2.1.2.30) and
the BLF indicators no longer function.
Has anyone had a similar issue? And a solution?
PaulH
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PaulH
On Wed, 2008-02-13 at 13:49 +1100, Mohammad Salaque wrote:
Dear all,
Anyone can point me how to soft hangup all channels using single
command ? I am using Asterisk 1.4.15.
thanks
Salaque
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Is there a way for Asterisk to generate anonymous SIP calls?
I have tried putting in the fromuser field, but that didn't seem to
work.
I have been asked by a Telco to provide SIP traffic with headers that
are empty before the '@' symbol.
Is this even possible?
PaulH
Astlinux on your own built box?
PaulH
On Thu, 2008-02-07 at 14:11 -0800, John Constalgie wrote:
Hi there,
I am looking to buy an Asterisk Appliance or Box for my organization
and was hoping to ask for recommendations.
My ideal box is a small device in size like Digium's AA50 Asterisk
I made up some dialplan rules to strip the '+' and replace with the
00...
Something like:
exten = _+XX.,1,Dial(zap/g1/00${EXTEN:1})
PaulH
On Thu, 2008-02-07 at 00:18 +, Ed W wrote:
Can someone please explain how to match a + character in a dial plan (so
that I can swap it for the 00
With some of the phones (snom, for example) you can turn off mwi, so the
phone will only accept one call at a time. Much easier.
PaulH
On Mon, 2008-02-04 at 19:27 +0100, Karsten Wemheuer wrote:
Hi,
I want to use GROUP_COUNT to limit calls to a specific destination. From
somewhere on the
I thought sounding like a dalek was a good thing.
PaulH
On Sun, 2008-02-03 at 23:56 +, Thomas Kenyon wrote:
I need to set up the sound card of a server to use in an overhead paging
system, as normal I am testing this on my home machine first (which has
slightly different Hardware).
Hi,
Does anyone of you has a working configuration with SNOM phones that are
able to pickup a call from a flasing LED?
Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 release, and
therefore I don't think any config changes will fix it. We've been told to
roll back to our
I need to carry a variable over into the 'h' priority - so I can go back
and clean up DB entries in a mysql database (time of call and so on)
I tried using UNIQUEID but it seems that 'h' generates a new one.
Anyone have any ideas? What can I use to carry a variable over into
'h'??
You might need Voicemailmain([EMAIL PROTECTED])
PaulH
On Thu, 2008-01-31 at 00:30 -0500, John Von Essen wrote:
Ok, I have spent all night trying to figure this out, and hopefully
somebody has a similar experience.
I have a very basic asterisk config. Sample configs, with the only
You can also look at routing based on number ranges (if you keep the
separate numbers in separate number ranges) but I would guess that this
is not going to suit your needs.
Maybe storing all the accounts in mysql (realtime) would also be a good
planh.
PaulH
On Wed, 2008-01-30 at
It doesn't actually work at all - I tried, and even logged a bug with
digium with no luck. :(
Are the queue logs not quite good enough?
PaulH
On Tue, 2008-01-29 at 17:20 -0800, Johnny Tam wrote:
How to make ${ANSWEREDTIME} to work with Queue, so when the user hangs
up, I can calculate how
Does turning off the notransfer help? I would imagine that dropping the
second server out of the equation might be useful, and save some
bandwidth.
PaulH
On Tue, 2008-01-29 at 10:38 +1100, Daniel Cole wrote:
Hello List,
I am currently having a bit of a strange issue with a pair of
What does 'show agents' give you? 'show queues' would be useful too.
PaulH
On Mon, 2008-01-28 at 13:36 +0100, Thomas Kenner wrote:
Hi,
when I'm trying to call the following extension
exten = 6002,1,Verbose(1|Extension 6002)
exten = 6002,n,Dial(Agent/6002)
exten = 6002,n,Hangup()
There was a cool paper written a a few months ago where they tested some
older dell servers - full details of specs and tests were available.
PaulH
On Thu, 2008-01-24 at 08:54 +1100, Daniel Cole wrote:
Sorry to be a little OT.. But may I ask what some more of the specs
are for that
On Wed, 2008-01-23 at 18:23 +, Gordon Henderson wrote:
Is there any way to find-out the peak number of calls that an asterisk
system has had? Not the total number of calls, but the maximum number of
simultaneous calls.
I know I can porobably go through the CDR logs and look for calls
On Wed, 2008-01-23 at 09:39 +0100, Alberto Pastore wrote:
Hi everybody.
I know maybe this question has been posted some time ago, but
I need your updated opinion on the subject.
I'm replacing our old pbx with asterisk.
I have two TE207 dual pri (e1) cards on a clustered system
(one on
, 2008 6:08 PM, Paul Hales [EMAIL PROTECTED] wrote:
There was a cool paper written a a few months ago where they tested some
older dell servers - full details of specs and tests were available.
PaulH
On Thu, 2008-01-24 at 08:54 +1100, Daniel Cole wrote:
Sorry to be a little OT
I love writing dialplan, using vi.
Does that make me weird?
PaulH
On Wed, 2008-01-23 at 23:57 -0500, Ken D'Ambrosio wrote:
Hi, all. I've done some Asterisk recelling, but recently got roped into a
Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much
all circuit-based
With comments like that people are going to think that we aren't
related.
PaulH
On Thu, 2008-01-24 at 16:46 +1100, Rob Hillis wrote:
Yes, but I already knew that. :)
Paul Hales wrote:
I love writing dialplan, using vi.
Does that make me weird?
PaulH
On Wed, 2008-01-23
I once attended an office with such bad cabling that we put the switch
on top of the server and ran cables against the walls to prove that the
internal cabling was rotten.
PaulH
On Tue, 2008-01-22 at 12:26 +1100, Cameron Hissey wrote:
After changing all the networking and removing PoE, the
You would have to write an external app to create a call file after each
vm is left...probably doable (externap in voicemail.conf), probably
fiddly.
PaulH
On Tue, 2008-01-22 at 17:44 -0500, arkda wrote:
My guess is you want the server to call the user and play the
voicemail?
On Jan 14,
, as multiple extensions were disappearing and
reappearing all at once. i swapped that cable out late yesterday
afternoon and i will wait today and see how it goes.
Thanks everyone for continuing to help me with this issue!
Sincerely,
Cameron Hissey
On Jan 23, 2008 9:32 AM, Paul Hales
points, i
dont really know how VLANS and segmentation are going to go...
Thanks so much everyone for your support!
Sincerely,
Cameron Hissey
On Jan 21, 2008 4:13 PM, Paul Hales [EMAIL PROTECTED] wrote:
Forwarded Message
From: Paul Hales
The granstream gxp-2000 has the blf/line buttons but they are terrible
phones.
Am I missing any phones? Any other suggestions?
I have to agree with your point - the transfer on the Snom's is not good
if you have to juggle several calls. The Polycom transfer system is
probably the best,
Generally, E1 is pretty rock solid so my guess is more inside the
network.
We found an issue at a site a while ago which was pretty bad (calls
cutting off randomly) and we fixed it by disconnecting the voice and
data networks. We could have troubleshot it properly, but fitting an
extra network
Use the chanisavail to check that the SIP channels are clear, and set
reasonable 'qualify=' settings for them
PaulH
On Tue, 2008-01-15 at 23:20 +0100, Jaap Winius wrote:
Hi list,
My Asterisk v1.4 system now has two ISDN channels and two SIP
channels. The idea is to make a dialplan
On Tue, 2008-01-15 at 17:44 -0500, Andrew Joakimsen wrote:
On Jan 14, 2008 6:29 PM, Paul Hales [EMAIL PROTECTED] wrote:
The 'setcallerpres' application is the one to use...
Only works for PRI channels (maybe plain T1) channels via Zaptel.
Agreed entirely.
PaulH
The 'setcallerpres' application is the one to use...
PaulH
On Mon, 2008-01-14 at 17:18 -0500, J. Oquendo wrote:
Hey all, when you guys have requests from clients to block their CID
from showing through, what are others doing? I had a coworker throw in
some Name Here0 garbage which none my
Maybe ADDQUEUEMEMBER(queue1|${CALLERID(num)) is closer to what you are
looking foror ADDQUEUEMEMBER(queue1) - without the pipe...
PaulH
On Sun, 2008-01-13 at 18:17 +0100, Stefan Guenther wrote:
Paul wrote
;Pause/unpause Queue
exten = 424,1,PauseQueueMember(|SIP/${CALLERID(num
Is there a good way to set the callerid(name) for calls being returned
from parking? We tried using the parkandannounce function, but we
couldn't get the audio to play back nicely. (we don't want the park
position returned as a separate phone call...)
ideas?
PaulH
What zap driver are you using? ztdummy?
PaulH
On Wed, 2008-01-09 at 15:41 -0500, Mike Coakley wrote:
I'm setting up a new Asterisk system on a Dell server and I'm getting
fuzzy voice between the Polycom IP SoundStation 550 and the Asterisk
server. I've checked all of my codec settings
Are they expecting numbers in a 61 format?
PaulH
On Fri, 2008-01-11 at 16:27 +1100, Kev S wrote:
Hi everyone,
having a issue with asterisk and my new Voip providers service.
Iv set up many asterisk systems before but never seen this and have
tried to fix this with no luck..
I
My question would actually be - is there any support for h234 over ISDN?
PaulH
On Mon, 2008-01-07 at 19:59 +0100, Olivier wrote:
Hi,
Asterisk now supports h234m.
Does anyone know a Media gateway such as those of Mediatrix, Patton,
Audiocodes, Cisco that also supports h324m flows ?
Then it's time to build zaptel, then rebuild asterisk
later,
PaulH
On Tue, 2008-01-08 at 13:16 +0800, Nhadie wrote:
Hi Matt,
it seems i don't have that command.
*CLI zap show channels
No such command 'zap' (type 'help' for help)
*CLI
! abort add ael
I like my Aastra 480i a lot. It has buttons with numbers on them.
What I am guessing you are really asking is 'what's a really good phone
that's really cheap'.
PaulH
On Fri, 2007-08-31 at 14:11 -0400, William Herrera wrote:
I need to quote a client for a job and I was just wondering.
Out
bad advice
that I think to be good. Yeah i'll be quite now.
On Jan 3, 2008 10:19 PM, Paul Hales [EMAIL PROTECTED] wrote:
Asterisk doesn't support g728.
Any idea what does?
PaulH
On Thu, 2008-01-03 at 20:57
Playback your message first, then call Voicemail with the 's' option?
PaulH
On Mon, 2008-01-07 at 13:14 +1100, Daniel Cole wrote:
Hello List,
I have a client (a nursing home) that we are looking at installing a
trixbox for. One of the features that they would really like is a
Some of the grandstream phones refuse to listen to Asterisk, so you have
to set them manuallygr.
PaulH
On Thu, 2008-01-03 at 17:23 -0400, William Herrera wrote:
The created extension its set to default (rfc2833). This is something I have
never had the need to change ... (with the
Asterisk doesn't support g728.
Any idea what does?
PaulH
On Thu, 2008-01-03 at 20:57 -0600, Kerry S wrote:
nothing? :'(
On Dec 25, 2007 1:59 AM, Kerry S [EMAIL PROTECTED] wrote:
Unfortunately I don't have a server set up that supports
G.728.
I'm
I put something together like this for a finance company - Asterisk
looked up the callerid in a MySQL database, and put the call into a
queue, with a higher priority if the call was from certain clients.
If the callerid was not found, it them allowed for a clientid and
pincode to be entered.
Creative use of the 'read' application?
PaulH
On Wed, 2007-12-19 at 17:12 -0800, Justin Killen wrote:
I have an application where a call-in user is prompted to enter an
identification number for schedule information. That id number is
setup as an extension, and if that extension doesn’t
You nailed it Randy!
When an Asterisk appliance and associated phones can compete with a
Panasonic KXTG-4000 (or similar) on terms including price, ease of use
reliabilitythat's when Asterisk for every grandma, aunt, uncle
counsins (who never finished high school) will be viable for
On Fri, 2007-12-14 at 11:33 +0100, Gergo Csibra wrote:
Friday, December 14, 2007, 5:47:38 AM, Paul wrote:
Umm - you could just buy a SPA-3000/3102/3666/etc.
What is SPA-3666?
The special red model.
PaulH
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Umm - you could just buy a SPA-3000/3102/3666/etc.
PaulH
On Fri, 2007-12-14 at 05:36 +0100, Vincent wrote:
On Thu, 13 Dec 2007 22:21:50 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
It is likely to be a very strenuous job to port the framework and all of the
drivers.
Too bad, because
There is a 5.2.2 firmware available now, but the changelog for it isn't
helpful at all.
PaulH
On Wed, 2007-12-12 at 11:22 -0700, Marty Mastera wrote:
We are having an issue with the SPA962/932 combo where the phone and
the sidecar will reboot unexpectedly – could be onhook, could be on a
Using the 'read' function you should be able to do something similar-
use the 'read' function to grab what you need, then push it into
accountcode.
From memory, I did something similar once.
PaulH
On Wed, 2007-12-12 at 12:32 -0500, Glenn Cobb wrote:
Greetings list...
I have a Trixbox
We are looking for an Asterisk tech in the Brisbane area...ideas?
PaulH
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What codec are you using?
PaulH
On Wed, 2007-12-12 at 13:00 +1100, Daniel Cole wrote:
Hello Everyone,
We have recently installed a pair of Trixbox servers in for a client
of our. They have two locations, with one server each. The servers
terminate 3 standard POTS lines into a Sangoma
'iax2 show channels'maybe
I have a feeling this is going to be one of those ugly ones where it's
going to be a pain to troubleshoot...
Offhand - have you tested 'trunk=yes' vs 'trunk=no'?
PaulH
On Wed, 2007-12-12 at 17:00 +1100, Daniel Cole wrote:
Hi Paul,
Where abouts exactly
I have seen a beta-level unit that also supported POE.
With regards to non-beta hardware, standard analog doorphones work
pretty well with Linksys SPA units.
PaulH
On Tue, 2007-12-04 at 02:53 -0500, Nick Seraphin wrote:
On a similar note... has anyone ever seen a SIP-based door intercom
I don't think it ever gets that cold here in Australia.
PaulH
On Wed, 2007-12-05 at 21:24 -0500, Jon Pounder wrote:
Quoting Paul Hales [EMAIL PROTECTED]:
another option is use some sort of linux based device n770, or even an
nslu2, and program a sip client to behave however you like
Yes - tomorrow night is the monthly Asterisk meeting held by the
Melbourne Asterisk group. (Melbourne, Australia that is)
Venue is usually Pint on Punt (corner of Punt Rd and Peel Street) from
7pm onwards.
Feel free to turn up, eat food, drink beverages and talk about Asterisk.
later,
On Sat, 2007-12-01 at 10:22 -0700, Anthony Francis wrote:
[EMAIL PROTECTED] wrote:
Anyone have an idea how to implement a phone number that can only be
called once? The first time it will process normally and any
subsequent calls will be rejected.
I think the newer version of the firmware fixes this problem.
Paul Hales
AsteriskIT
On Tue, 2007-12-04 at 00:13 -0400, Doug Meredith wrote:
I have searched for this without much luck. I want to be able to send
public-address-like notices over VoIP phones. The LinkSys SPA-941
auto-answer
to be seen by the card to the Dictaphone
channels.
Thanks in advance for any suggestions.
Regards,
Paul
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The current Digium BRI cards need the phones to send DTMF over as
SIP-INFO.
Not sure why, but googling should help. (I think this is even covered on
the Digium site)
PaulH
On Wed, 2007-11-28 at 09:47 +0100, Administrator TOOTAI wrote:
Good day all,
we have following setup: Debian Etch 64,
I also understand your stand here Kevin - there is no way you can
restrict the software running on a server out in the wild, and no way to
make sure the software they are running will not conflict in any way.
But a single port E1 card with hardware echo cancellationpossible?
PaulH
On Wed,
From memory - 'rtcachefriends=yes' should do the trick.
PaulH
On Wed, 2007-11-28 at 16:56 -0800, Daniel Hazelbaker wrote:
I am trying to get the presence/hints/BLF working along with Realtime
SIP but I never get any busy notification. core show hints always
shows the realtime sip user
I caught the wrong bus once, and ended up as part of a murder
investigation.
Let this be a lesson to everyone!
PaulH
On Wed, 2007-11-28 at 06:50 +, broadband Voice wrote:
Hi,
Can anyone assist me in resolving this problem? I installed the G729
on a 32 and just found out that the
Do the SIP-FXO gateway devices do any better?
Eric ManxPower Wieling wrote:
Asterisk does not detect analog ports with no line plugged in. It does
not test for dialtone before dialing (this applies to all analog cards
except the X100P).
Rilawich Ango wrote:
It works if it specified the
this will be fixed in a firmware update in the near future.
cheers,
Paul.
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Hey, we are looking for someone to work to the end of january , and
maybe even stay on after that.
_Immediate start_.
Low to Mid level asterisk work (phone support and onsite install work)
You MUST be living in Melbourne, Australia.
Email me off list for more details.
PaulH
The dialplan command 'setcallerpres' is also good.
PaulH
On Fri, 2007-11-23 at 12:44 +1100, Nick Brown wrote:
You can set callerid within the [general] section of your sip.conf.
This should work for you.
On 23/11/07 8:02 AM, Mike [EMAIL PROTECTED] wrote:
Hi,
All looks fine to me - and hopefully nobody does anything nasty to your
server.
PaulH
On Thu, 2007-11-15 at 06:15 +0800, Richard Cahilig wrote:
Hi,
I installed asterisk-addons and asterisk-stats, Its working now except
of one problem. The problem is there is no call logs when you open the
to try and
reproduce it, plus it would be a purely academic project as if there
was a bug it has since been fixed.
Thanks for the suggestions Paul.
Nick.
On 13/11/07 4:48 PM, Paul Hales wrote:
Is it possibly a funny zaptel issue? Paul Hales AsteriskIT On
Tue, 2007-11-13
What format is your music on hold in?
PaulH
On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote:
Afternoon All,
Today rolled a pre-production box from Trunk back to 1.4.7 (In an
attempt to get a working SCCP channel). During the process Music On
Hold appears to have died (Not, just when
http://www.oldskoolphreak.com/tfiles/voip/chatter_bug.pdf
PaulH
On Mon, 2007-11-12 at 21:07 -0800, Robert Goodyear wrote:
Does anyone know anything about the Chatterbug product? I can't tell
if it's an ATA with a modem or some sort of LCR proxy or somesuch.
Anyone?
Is it possibly a funny zaptel issue?
Paul Hales
AsteriskIT
On Tue, 2007-11-13 at 16:20 +1100, Nick Brown wrote:
It was using the 3 wav's from Freeplay. I have just recompiled and told it
to pull down the ULAW versions, then removed the Wav's however it has made
no difference.
Cheers
Hi Dan,
Thank you for your answer. I am using asterisk 1.4.13 and keepalive has a
value of 120 in skinny.conf.
2007/11/8, Dan Austin [EMAIL PROTECTED]:
Paul wrote:
I have six cisco 7911g connected on asterisk over
chan_skinny. Four of them are working OK. two of
them even the screen
I have found the new 7.x.x series firmware to be pretty much unusable in
speakerphone mode, which is slightly disappointing as I like the Snom
phones.
PaulH
On Fri, 2007-11-09 at 03:34 +0100, Philipp Kempgen wrote:
Jason White wrote:
On Thu, Nov 08, 2007 at 10:22:41AM +0100, voip crazy
I managed to use Cisco IP phones 7911g with asterisk with Sccp and
chan_skinny without any configuration files in tftp. Only settings in
dhcpd to indicate the tftp address and skinny.conf settings. the problem
that I have is that from 8 phones two of them after working a while now are
Hi everybody,
I have six cisco 7911g connected on asterisk over chan_skinny. Four of them
are working OK. two of them even the screen on the phone is indicating that
is registered and has number loose connection to asterisk . On asterisk the
message is Skinny Client was lost, unregistering.
My memory tells me that there is a flag (something like 'ringinuse')
which can make sure this sort of thing does not happen.
PaulH
On Mon, 2007-11-05 at 10:26 +1100, Nick Brown wrote:
Morning All,
Quick question that has me stumped. Have a queue with several members
(Statically defined in
on the
Asterisk box, to act as a T.38 endpoint. This appears to be the result
of a licensing issue with SpanDSP.
http://www.voip-info.org/wiki/view/T.38
That's a real shame as T.38 termination support is one of the last big
pieces for us to make Asterisk a seamless solution.
Paul Bryson
[EMAIL PROTECTED] wrote:
I thought there was some talk of getting T38Gateway into asterisk_addons?
Stupid linking bullshits.
Stupid indeed. I'm surprised T.38 support isn't a higher priority for
Digium, given that faxing has such a high failure rate with VoIP.
Paul
for address/port to
send to
set_destination: set destination to 10.0.2.136, port 5060
Reliably Transmitting (no NAT) to 10.0.2.136:5060:
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.4.147:5060;branch=z9hG4bK4c02bc4f;rport
From: Paul sip:[EMAIL PROTECTED];tag=as1ed7b694
To:
sip
gateways?
Regards,
Paul
This e-mail is intended solely for the addressee and is strictly confidential;
if you
And
Insecure=very
Yet I can ONLY get 407 Proxy Authentication required.
Can anyone give me even a hint in the right direction?
Thanks
Paul
, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=277024dd
Content-Length: 0
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Campbell
Sent: 26 October 2007 13
over tftp . I am using
asterisk 1.4.13.
Paul
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results in a 407 Proxy Authorisation Required error on Asterisk and a
bridge_reject noautho
The server is registering fine with Asterisk.
Some debug stuff:
=
sip.conf
[general]
context=internal
srvlookup=yes
[paul]
type=friend
secret=removed
qualify
.38 data.
6. Exchange 2007 converts the T.38 data to an image in an email and
stores it in the user's inbox within Exchange.
Paul Bryson
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We had issues with TE110p cards in Dell 860's, but TE120p's fixed the
problem.
PaulH
On Wed, 2007-10-24 at 21:12 -0400, Joseph Begumisa wrote:
Has anyone had any compatibility issues with a TE110P card installed
on a Dell Poweredge 1950? I noted the following error on the LCD
display
We have written stuff previously for most major phones that does
auto-deploymentserver sits there waiting for phone to ask for
configs, when the phones hit the server, the configs are written on the
fly.
Bit fiddly to write, but once it's going it's pretty good.
PaulH
On Sat, 2007-10-20
What we found is that even if you get the lights working, they go off
after a few days.
Paul Hales
AsteriskIT
On Mon, 2007-10-22 at 09:49 -0300, Carlos Maimone wrote:
Dear friends,
I am working around with a Snom 360 and Asterisk 1.4 + FreePBX
In order to get subscriptions working
The Xorcom Astribanks are quote good - have you looked at those?
PaulH
On Tue, 2007-10-23 at 12:41 +0800, Rilawich Ango wrote:
What do you mean by interruption? Is it possible to better control to
prevent it? The options you provided is over my budget. That's why
I am looking for
Agreed - handling multiple calls and transferring them on a Snom is a
problem. Too fiddly.
Polycom phones work well in reception situations, if set up well.
Haven't tested the new Aastra's (but the Aastra transfer function works
well) but they would probably be OK too.
PaulH
On Fri,
I know of a call centre that bought a cheap projector for that purpose.
PaulH
On Thu, 2007-10-18 at 23:28 -0700, o o wrote:
Has anyone used an LED wall display with asterisk? I have a customer
who has an ancient telecorp system that drives an LED wall display. It
shows the number of agents
adding a service
provider foo.bar.com and a calling rule to send all calls for
extension 6002 to that provider, but all I get is Service unavailable.
With the Asterisk-docs site down I'm finding it tough going.
Thanks for any pointers you can give me.
Paul Campbell
On Wed, 2007-10-17 at 14:53 -0700, shadowym wrote:
Ok Thanks,
I guess I'll have to give it a shot. I just assumed it would be more work
than 30minutes (after the initial learning curve) for a moderately complex
dialplan..
The other issue that arrives is that a complex dialplan can't be
I use vi. Not sure if it has a web interface yet.
PaulH
On Tue, 2007-10-16 at 00:51 +0200, Dovid B wrote:
None. Asterisk vanilla is the best IMHO.
- Original Message -
From: Anciso, Roy
To: asterisk-users@lists.digium.com
Sent: Monday, October
Setting up a static queue (with sip members) is generally the best way
to do this.
That way, the dialplan simply has a line like
exten = s,2,queue(ringall|i||300)
PaulH
On Tue, 2007-10-16 at 23:02 -0400, Rich wrote:
Asterisk 1.4.2
I have spent much of today trying to make a DID (from SIP
Try the Prescott version of the G729 .so.
That one is made for xeon's.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Lynchfield
Sent: Friday, October 12, 2007 2:36 PM
To: Asterisk Users Mailing List - Non-Commercial
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