Hi,
is the support of real-time text limited to the SIP channel driver only?
Somehow Asterisk is not offering T.140 to the called party when
initiating a call and including real-time text.
In my pjsip.conf I allowed T.140 and enabled text support.
Regards,
Simon Hohberg
ion not found in context 'from-extern'.
What am I doing wrong?
Regards,
Simon
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: ht
106>;tag=m6bqn333dr
To: <sip:6000@192.168.2.106>;tag=as1792125e
Call-ID: 1ansppdrpdulbtr3j5ub
CSeq: 6409 BYE
Server: Asterisk PBX 12.8.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, M
On 06/27/2016 12:09 PM, Joshua Colp wrote:
Simon Hohberg wrote:
Hi,
I want to pass a part of a SIP INVITE multipart body. I found a quite
old patch here:
https://issues.asterisk.org/jira/browse/ASTERISK-14510?jql=text%20~%20%22body%20part%22
But this patch is for the SIP channel driver
to put in the
dialplan then?
Thanks in advance,
Simon
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http
that loads the page), which probably makes sense
only for development. Otherwise you have to use https and wss for the
reasons discussed earlier.
Hope it helps.
Simon
--
_
-- Bandwidth and Colocation Provided by http://www.api
rial+using+SIPML5
[2] https://www.doubango.org/sipml5/
Regards,
Simon
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
h
the first RTCP Full
Intraframe Request (FIR) and the received video stream cannot be
rendered till the next FIR 90s later arrives.
Am I right or is this nonsense?
Is this a known issue? I couldn't find anything about this.
Is there a fix available?
Thanks in advance!
Simon
Anyone?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To
Hi,
I am starting a new project to develop a predictive dialler system.
- Agents can start receiving calls from the queue if agent press
Available button on the browser which will unpause the queue on Asterisk.
- About 100-150 concurrents calls on a Asterisk box
- Call-out initiated. Other end
Hi all,
We are running a fax2email service based on asterisk 1.8.18.0, and we are
currently trying out asterisk 1.8.32.2 in our labs. We get the following
error when sending faxes out:
[Apr 7 14:34:20] ERROR[16653]: res_fax.c:2121 sendfax_exec: 'modems'
setting 'V17,V27,V29' is incompatible with
context? I've heard that it's better to wait a few
seconds before calling ReceiveFAX(), is it still necessary in case we don't
actually need fax detection?
Simon
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Hello,
I have a Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] card which
seems to work on 1 machine but not on another. It SHOULD load this driver:
dahdi_hardware -v
pci::00:00.0 wcb4xxp+ 1397:08b4 Junghanns QuadBRI ISDN card
Instead of:
dahdi_hardware -v
, but Asterisk is much better suited to handling
end-user devices. The external service does control logic only.
Can someone point me at the right place in the documentation to get a
handle on where I should be hooking things like this?
--
Cheers
Simon
On Wed, 10 Apr 2013, Simon Green wrote:
Hi there, I’m new to Asterisk and there’s a ton of documentation. I’m not
really sure where to start. What I want to do is this: a PBX service ala
FreePBX, but where call control is passed via SIP to an external service
which will tell Asterisk
It's nearly there now, just need a few more votes in order for it to
trigger the next phase. Please take a moment to vote if you're
interested:
http://area51.stackexchange.com/proposals/12932/telephony/
On Mon, 9 May 2011, Simon P. Ditner wrote:
For those of that are fans
For those of that are fans of stackoverflow.com, and stackexchange.com,
there's an effort to define a telephony stackexchange site. It's still in
the definition phase. What it needs to move forwards is more votes on
on/off topic questions, and perhaps some better questions to vote for or
Hi Asterisks Team,
I am getting the error below after getting a connection to a telco using
ss7. Anyone know how to solve it?
The link keeps coming up and down every 30 seconds.
Resetting CIC 3
[Mar 28 15:16:28] WARNING[20434]: chan_dahdi.c:11913 ss7_linkset: RSC on
unconfigured CIC 3
Received
Hi,
I am trying to set up asterisk with ss7. Whenever I try to load module
chan_dahdi.so, I get the error
[Mar 26 17:33:27] ERROR[10437]: chan_dahdi.c:10458 mkintf: Unable to find
linkset -1
I have compiled dahdi, libss7, asterisks (am using asterisk 1.6) in that
order. Have already set
and they are using something somewhat standard
like qemu.
Cheers,
spd
--
| It ain't what you don't know that gets you into trouble. It's what
| you know for sure that just ain't so. -- Mark Twain
|
| Network: http://www.linkedin.com/in/spditner
| http://facebook.com/people/Simon-P-Ditner/776370031
driver with the B410P, please comment out the wcb4xxp line in
/etc/dahdi/modules.
This will prevent DAHDI from loading wcb4xxp which will conflict with the mISDN
driver.
Enough reading.. if you're still awake! Any help would be very much
appreciated.
Thank you,
Simon
Date: Wed
Hi folks,
I wonder if any of you out there are using Siemens S685IP base station(s) (with
S68H handsets) on Asterisk and experiencing problems with SIP registrations
where the SIP extensions do not ring and peers become unreachable after a
period of time.
Symptoms are rather sporadic,
# voip
sip.conf:
[101]
callerid=Simon Tennant
type=friend
username=101
secret=xx
host=dynamic
reinvite=no
canreinvite=no
mailbox=101
context=from-internal
nat=yes
port=5060
qualify=yes
insecure=very
disallow=all
allow=alaw
also sip.conf
[justvoip.com]
type=peer
host=sip.justvoip.com
back at http://taug.ca for event updates.
Cheers,
Simon P. Ditner
TAUG.ca Talk Coordinator
[1] http://www.sangoma.com
[2] http://www.palosanto.com
[3] http://www.elastix.org
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Hi there,
Is anyone using a headset with one of these phones? If so, can you
recommend any?
Thanks
Simon
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http
So any 2.5 headset will work with the SPA922?
On Fri, Aug 1, 2008 at 12:23 PM, Paul Hales [EMAIL PROTECTED] wrote:
Plantronics.
PaulH
Simon wrote:
Hi there,
Is anyone using a headset with one of these phones? If so, can you
recommend any?
Thanks
Simon
Hi There,
Is there a way just to have the custom voice message play, and not
have asterisk play: vm-intro after that?
Thanks
Simon
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix
Got it!... we are using Elastix and i just had to set s in the
VM_OPTS in the extensions_additional.conf file.
On Sun, Jul 20, 2008 at 1:41 PM, Simon [EMAIL PROTECTED] wrote:
Hi There,
Is there a way just to have the custom voice message play, and not
have asterisk play: vm-intro after
software/config wise to help with this?
I did find this (http://www.misdn.org/index.php/FAQ):
4) You set another value for tx-gain, -1 for example to prevent
echoes. Please set the tx-gain back to 0 for those calls as in 3)
(vt0).
Can anyone comment on this please?
Thanks
Simon
/10.3... will openSUSE
11 work here?
Simon
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE
On Thu, Jul 3, 2008 at 5:07 PM, Dave Cotton [EMAIL PROTECTED] wrote:
Simon wrote:
Hi There,
Has anyone managed to get 2 AVM ISDN Fritzcard's working in with a 2.6
kernel system?
Yes, with Suse 10.2/10.3 and chan_misdn.
OK. ive got debian etch working with one card compiling the drivers
Hi There,
Has anyone managed to get 2 AVM ISDN Fritzcard's working in with a 2.6
kernel system?
Simon
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http
to start here.
Thanks
Simon
asterisk:~# capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.11-07 (49.23)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
internal controller supported
DTMF supported
Supplementary
Hi There,
I am looking to build an Asterisk server with dual AVM Fritz!PCI cards
linked to 2 BRI in New Zealand. Just wondering if anyone has done
this, and if you have any ideas about the best disto choice for this
task?
Simon
___
-- Bandwidth
What open source tools are people using to quantitatively measure how
well QoS/traffic shaping is performing out in the field, and what call
quality people are experiencing in terms of jitter and packet loss?
Cheers,
spd
___
-- Bandwidth and
Cool - thanks Rob. I will check it out tmorrow.
Simon
On Wed, Apr 16, 2008 at 4:34 PM, Rob Hillis [EMAIL PROTECTED] wrote:
IIRC Asterisk doesn't support the full presence publishing spec so you
won't get the full range of possible status types, however you should at
least get free/busy. I
, Med, Low settings for: FTP, HTTP, Telnet, SMTP
and POP3. Plus we have the ability to specify up to 3 ports for the
same settings.
Is this worth doing? If so, what ports should i specifiy?
Simon
___
-- Bandwidth and Colocation Provided by http://www.api
Hi There,
We have some users using x-lite as their SIP phone... but im wondering
how to get the Calls Contacts to show as being available (Or if it
can be done at all?). Is this what Presence is?
Thanks
Simon
___
-- Bandwidth and Colocation Provided
for agents!
Simon wrote:
Hi There,
We have some users using x-lite as their SIP phone... but im wondering
how to get the Calls Contacts to show as being available (Or if it
can be done at all?). Is this what Presence is?
Thanks
Simon
Thanks again!.. Right. I have it working now, it shows the users
statuses as online or offline and changes them when someone closes
their app. But not free/busy type changes.. Any idea why here?
Simon
On Wed, Apr 16, 2008 at 3:21 PM, Rob Hillis [EMAIL PROTECTED] wrote:
X-Lite. Of course
incorrect here?
Thanks
Simon
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
You could try:
asterisk -rx database get SIP/Registry 101 | cut -f 2 -d ':'
Which is not much shorter, but probably more efficient
Simon Elliston Ball
[EMAIL PROTECTED]
http://www.simonellistonball.com/
On 31 Mar 2008, at 10:02, Olivier wrote:
Hi,
Sometimes, you need to send requests
,
but it's probably a lot easier to just use the registry database, just
depends on how often you're going to be doing the lookups.
simon
Simon Elliston Ball
[EMAIL PROTECTED]
http://www.simonellistonball.com/
On 31 Mar 2008, at 10:56, Olivier wrote:
2008/3/31, Simon Elliston Ball [EMAIL
) meaning as far as the server is concerned
the IP communication is going down the same logical connection between
VOIP adapter and Asterisk server.
Has anyone else seen this and
is there either a work around or fix?
Many thanks
Simon
--
Simon Falvey
http://www.digium.com/en/products/voice/tc400b.php
Simon Elliston Ball
[EMAIL PROTECTED]
On 1 Feb 2008, at 17:29, Charles Feng wrote:
Hello All:
Does the Asterisk support to insert an off the board transcoder
for a call?
Thanks,
Charles
Never miss a thing. Make Yahoo your
might have to check out the
biz edition too. It's all looking good. Good luck with the next release!
Simon
Simon Elliston Ball
[EMAIL PROTECTED]
On 23 Jan 2008, at 08:35, Zoa wrote:
You can find it here:
http://www.zoiper.com/downloads/free/linux/zoiper201-linux.tar.gz
Note
Looks interesting. I couldn't get it working because a few of the
preference fields were not responding (current svn, build on Leopard).
Looks like a nice elegant solution though. Let me know if there's
anything you want help on and I'll dust off my cocoa!
Simon
Simon Elliston Ball
[EMAIL
Try:
http://www.voip-info.org/wiki/view/Mysql
and the links thereon.
simon
Simon Elliston Ball
[EMAIL PROTECTED]
On 16 Jan 2008, at 19:11, Naveen Palani wrote:
Hello,
Is there a possibility to connect from asterisk to mysql database
without the interface application like Ruby or PHP
a reasonably low qualify value.
Simon
Simon Elliston Ball
[EMAIL PROTECTED]
On 11 Dec 2007, at 15:15, asterisk wrote:
I don't know of a way without reloading. Realtime still needs a sip
reload.
Look at the dial command. There are options that you can add that
will
disable re-invites per
?
If anyone is, what's the preferred way to keep timing accurate?
Thinking I may have been too hasty in switching to UML...
S.
--
Simon Tennant ___ http://imaginator.com/~simon/contact
signature.asc
Description: OpenPGP digital signature
___
Sign up
= s,3,WaitExten()
The ringing sound doesn't work for any extension if I use this one. I just get
silence until someone answers. How come?
I use Asterisk 1.4.10. I have attached my extensions.conf file to this email.
Thanks,
Simon
[globals]
SIPTRUNK=418555
IAXTRUNK=514555
[default
On Wednesday 29 August 2007 10:46:18 Eric ManxPower Wieling wrote:
You do not have a /etc/asterisk/indications.conf This file is used to
provide ringing sounds AFTER a channel has been answered.
Thanks a million times!
___
--Bandwidth and Colocation
. That's
all. Hope it's helping.
Adi.
On 6/1/07, Luis Morales [EMAIL PROTECTED] wrote:
Hi Adi,
My be better if you send us the code about how did you do to catch and
retrive the data from asterisk.
Regards,
Luis Morales
On Fri, 2007-06-01 at 01:21 +0300, Adi Simon wrote:
Hi Martin
this issue or can shed some light on the
module crossover.
For reference the issue happens with both kernels I have tried 2.6.20.4
and 2.6.21.3 and both asterisk 1.4.2 and 1.4.4.
Any help appreciated.
Regards
Simon Alman
___
--Bandwidth and Colocation
Hi,
I'm trying to find a way of passing the actual call duration (something like
ANSWEREDTIME) to an AGI
script that runs periodically during a call. Any ideas?
Thanks,
Adi.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
PROTECTED] *On Behalf Of *Adi Simon
*Sent:* Thursday, May 31, 2007 5:54 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Passing call duration to an AGI Script
Hi,
I'm trying to find a way of passing the actual call duration (something
like ANSWEREDTIME) to an AGI
script
no difference)
Regards
Simon
Salvatore Giudice wrote:
You should get a packet capture of both cisco-cisco and
grandstream/polycom-cisco. Compare the SDP's. The cisco phone may not be
able to understand the other vendor's devices. BTW, what version of firmware
are you running on the cisco phones
be appreciated. We are running an old Asterisk server
with version 1.0.10 (yeah we know) and the same mix of hardware and
configs works fine.
On the new (problem) setup we are running Asterisk 1.4.2 and our Cisco
firmware is 08-2-00.
Any help appreciated.
Regards
Simon Alman
zaphfc
I am using a billion hfc card
Any pointers?
--
Simon Faulkner 01538 303 900
Staffordshire Moorlands
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http
I am using a billion hfc card
apt-get install zaptel-source
m-a a-i zaptel
Precompiled zaptel drivers should hopefully be added soon to Unstable /
Testing .
Thank you :-)
___
--Bandwidth and Colocation provided by Easynews.com --
Hey all,
The Toronto AUG has been working with Clue.ca and IT360
(LinuxWorld/NetworkWorld), and has put together a mini-asterisk
conference within their larger conference:
http://www.it360.ca/asterisk.cfm
If you're interested, as an 'association' we get 25% off the listed
prices. Our dicount
as to the Warnings ? I'm not quite at the stage
where I can test my setup yet and wanted to check before I get there.
Many thanks for your time.
Simon
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
Cool, thanks for the info.
Simon
Doug Lytle wrote:
Simon Alman wrote:
Hi Guys
I'm configuring a TE212P card and have the following two entries in
my /etc/asterisk/zapata.conf
switchtype=dms100
signalling=pri_cpe
When I reload asterisk I get the following messages:
-- Reloading
the option of pushing # to terminate party 3
(should the call only reach party 3's voicemail).
Either that or a ways to do DISA from within the meet-me functionality.
I can't imagine I'm the only person with this sort of requirement.
--
Simon Tennant http://imaginator.com
'lo,
A provider sets up an Asterisk box in order to service the needs of a small
number of customers. The provider issues SIP handsets and the users
register with sip.telco.com
Thanks to the selection of a brilliant family of technologies, including SIP
and Asterisk, the telco.com company
limit : 0
Dynamic : Yes
Callerid : Simon Tennant (Nokia E61)
Expire : 3595
Insecure : no
Nat : Route
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Trust RPID : No
Send RPID: No
DTMFmode : auto
LastMsg : 0
of the SIP hardphones, they provide an
excellent protection to upload bandwidth. They also seem to do some
early dropping on incoming traffic to persuade the ISP's routers to
slow down downloads once a call has been going for a bit, hence they
can limit downloads as well.
simon
On 4 Jan
The Fedora Extras rpm is tiny because it has nothing really of help
in it. It's missing the modules.
I've had some success on Fedora Core 6 using the ATrpms repository,
which has the zaptel-kmdl package for most variations of kernels
included in FC6.
Simon
On 14 Dec 2006, at 22:31
/101-08186e70, ) in new stack
== Spawn extension (mainmenu, t, 2) exited non-zero on 'SIP/101-08186e70'
Where am I going wrong and do I need to worry about Sent into invalid
extension 's' in context 'mainmenu' on SIP/101-08186e70 warnings?
S.
--
Simon Tennant http
Doug Lytle wrote:
Doug Lytle wrote:
Simon Tennant wrote:
[internal-extensions]
exten = 100,1,Goto(mainmenu,s,10)
You can't start at 10 on your menu, you have to start with 1.
strange - I jumped into that context at 10 and numbered up from 10 - I
thought that was ok.
Also when I started
Hi,We can provide UK toll-free inbound and can recommend a co-lo provider where you'll have single-hop access to our network. Feel free to contact me off list.Kind regards,Simon
On 9/27/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Can anyone direct me to a colo provider in the UKwhere I can
Mainly I have a problem of figuring out how to use them with dispatcher
or any other mean of switching between asterisks. Do you have any configuration
example of such?
On 9/28/06, Simone Ricci [EMAIL PROTECTED] wrote:
Adi Simon ha scritto: Hi, Did anyone actually manage setting up a single SER
Hi,
Did anyone actually manage setting up a single SER with multiple Asterisk boxes?
I particulary have a problem of keeping the session alive and by that I mean directing
all the following sip messages to the same asterisk box the first signal was sent (randomally).
Please don't direct me to
,Zac Amsler, Network OperationsSur-Tel Communications, Inc. NetIQ Systems, LLC* US48, Canada, A-Z Wholesale Termination.* US48 Origination, Toll Free DIDs.* Toll Free Termination (FREE).
Adi Simon wrote: Hi, Did anyone actually manage setting up a single SER with multiple Asterisk boxes? I
Hi Doug,On 9/25/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Lets say you have a cluster of, say, 10 Asterisk servers. After doing a local lookup to see if a number is available locally, in order to find out if the number is available on one of the other 9 servers, this peer has to query all 9
Hi Ronald,On 9/19/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:
I am thinking if re-invite will interfere accounting.No it won't
Please help me to figure it out:Phone A is registered at asterisk and calls a gateway. If the gatewayallows re-invite than the rtp would go directly from phone A to
ZoIP 0.2.0, the Zork/Asterisk bridge has finally been released. Now you
too can play 80's era text adventures over the phone using
text-to-speech, and speech recognition ;-)
What's a text adventure like you ask? Well, depending on your skill, a
typical dialog might go something like this:
Hi Ben,Yes it is but you need to remember to still include[macro-stdpbx1exten]switch = Realtime/in your extensions.confSimonOn 9/6/06,
Benjamin Jacob [EMAIL PROTECTED] wrote:
Hello all,Another question related to Realtime.Is it possible to call macros using Realtime arch?I have a macro definition
Hi Steven,The provider's implementation will have a bigger affect than any differences within Asterisk, e.g. how they are load-balancing and whether in fact SIP is serviced by Asterisk at all. Compared like-for-like within Asterisk we find there is not a lot in it, with each having their own pros
that joined the conference.
Is there a way to store this in a variable before they join the
conference? Or perhaps a way to detect the last user to join the
conferences number?Any help is appreciated.Cheers,- Simon Austin
___
--Bandwidth and Colocation provided
the
conferences number?Cheers,Simon
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi Benny,The E61 handles this just fine. With SIP as the default channel to dial and no WiFi coverage, you get a message asking if you'd like to dial by cellular. Works nicely other than a few stability issues.
SimonOn 24 Aug 2006 11:23:39 +0200, Benny Amorsen [EMAIL PROTECTED] wrote:
H ==
Hi Haspers,Makes sure you have created an 'Internet tel' profile. It doesn't appear to do anything but was vital in getting it working for me. The other settings in the how to look sensible.Simon
On 8/24/06, Haspers [EMAIL PROTECTED] wrote:
Strange,What settings do you use? I followed this
Hi Andreas,That is incorrect. It works just fine through NAT providing:- The server is proxying RTP as it has no support for STUN etc.- The NAT is the basic domestic router style, not a full blown firewall requiring port mappings
SimonOn 8/24/06, Andreas Sikkema [EMAIL PROTECTED] wrote:
Anyone
Hi Andreas,I'm on 1.0610.04.04 19-04-06 RM-89WOn 8/24/06, Andreas Sikkema [EMAIL PROTECTED]
wrote:Simon, That is incorrect. It works just fine through NAT providing:
- The server is proxying RTP as it has no support for STUN etc. - The NAT is the basic domestic router style, not a full blown
?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Simon
WoodheadSent: donderdag 24 augustus 2006 12:51To: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[asterisk-users] Re: Nokia E60/61/70 and SIP
Hi Haspers,Makes sure you have created an 'Internet tel
Hi Doug,We run with SOAP on both the client/server side, i.e. conventional AGI making SOAP request to SOAP server running on Apache. I'm sure FastAGI would be quicker but wasn't around and other than the overhead of making the SOAP request is very lightweight on the Asterisk side.
SimonOn 8/22/06,
We've done this with OpenVPN and it works fine. I'd recommend that the VPN server is not on the same box as Asterisk. Stick it on a firewall/gateway box giving access to the network containing the Asterisk boxes behind it. This way the Asterisk box(es) is seeing normal unencrypted traffic and the
note http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions
comment from Philipp Dunkel.
On 22 Aug 2006, at 17:13, Douglas Garstang wrote:
-Original Message-
From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 22, 2006 9:45 AM
To: Asterisk Users Mailing
sip settings) which are then joined together
in various ways to produce views for the extensions and a sipdevices.
Simon
On 22 Aug 2006, at 15:20, Douglas Garstang wrote:
The unofficial docs on the voip wiki for the realtime extensions
table structure is:
CREATE TABLE `extensions_table
no it doesn't. you could just change the context field for the
extensions you wanted to comment out.
On 22 Aug 2006, at 16:11, Douglas Garstang wrote:
Thanks, but that means I'd have to effectively comment out every
extension in that context, which isn't very feesible.
-Original
too much trouble from the srpm, but
it's a lot easier to stick to 2157.
If anyone else have managed to get FC5 to install the correct devel
packages for the latest kernel, please let me know!
Simon
On 21 Aug 2006, at 11:52, Tomislav Parčina wrote:
I'm trying to install Zaptel 1.2.7 on Fedora
In which case your best bet is probably to install with an rpm --
rebuilt on the source rpm.
simon
On 21 Aug 2006, at 12:36, Tomislav Parčina wrote:
In article 344F8B3D-6591-4001-9DE6-
[EMAIL PROTECTED], [EMAIL PROTECTED]
says...
I managed to get zaptel to compile reasonably easily
Hi,
I'm trying to accomplish having a single extension that always answers
with an automated voicemail prompt and record a user message, but can
recognize if the call is fax and handle it accordingly. Anyone here has any
experience with this kind of configuration?
Thanks,
Adi.
Hi Curt,That probably suggests that with SIP they're handing off the RTP to their upstream provider and just dealing with the signalling which is very low overhead. With IAX they have to transport both unless they're interconnecting upstream by IAX and can transfer. In my experience the load is
, Russell Bryant [EMAIL PROTECTED] wrote:
- Simon Austin [EMAIL PROTECTED] wrote: I have confirmed that GET VARIABLE doesn't return global variables in version 1.2.10 and submitted the following bug report:
http://bugs.digium.com/view.php?id=7609I'm not sure if you have seen it, but I posted
Thank you.I had previously seen the Local channel channel but I didn't completely understand how it worked. That was a really good explanation of how I can do it.Cheers,- Simon
On 7/27/06, Moises Silva [EMAIL PROTECTED] wrote:
may be im missing something, but i think the pseudo channel you
I have confirmed that GET VARIABLE doesn't return global variables in version 1.2.10 and submitted the following bug report:
http://bugs.digium.com/view.php?id=7609
Cheers,On 7/27/06, Russell Bryant [EMAIL PROTECTED] wrote:
On Thu, 2006-07-27 at 19:02 -0400, Simon Austin wrote: Is it possible
] wrote:
Worked on same version when I did it...using
PHP
- Original Message -
From:
Simon
Austin
To:
Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, July 28, 2006 3:52 PM
Subject: Re: [asterisk-users] accessing
dialplan global
channel that I've created to a meetme conf that plays a message? Does anyone know how to do this?
- Can I override the MOH and stream a recorded message into the conference with only the single user in the meetme conf?Any help/ideas are appreciated.Cheers,
- Simon
1 - 100 of 379 matches
Mail list logo
