On 4/7/2018 5:50 AM, Daniel Tryba wrote:
On Fri, Apr 06, 2018 at 02:27:31PM -0500, Brent Davidson wrote:
I have multiple Asterisk instances set up in different locations and would
like to modify the callerID of inbound calls to identify which instance the
call is coming from. I knew how to do
I have multiple Asterisk instances set up in different locations and
would like to modify the callerID of inbound calls to identify which
instance the call is coming from. I knew how to do that with the old
sip format, but can't seem to figure it out with PJSip.
For example:
Currently
as .au files and listen to them there are obvious points where
the audio just goes silent in the middle of the person speaking, and it
effects both directions. Doesn't make any sense.
On 4/4/2018 10:33 AM, Brent Davidson wrote:
At the first office, I replaced all the wiring except the in-wall
At the first office, I replaced all the wiring except the in-wall
stuff. Checked all the cables to make sure they were correct. I've
done cabling for the last 20+ years, so I've usually got a good feel for
that. Make all my own cables and do all my own wiring. I still make a
habit of checking
Well, I now have another office complaining of the audio drop-outs. Logs
are showing the same issues. RTP just stops for awhile then resumes.
At the original problem office, I replaced all the network cables,
replaced two network hubs, and made sure the phones are all connected
correctly.
I'm having a strange issue with Asterisk 13.17.2 and pjproject-2.7.
I have one extension that will occasionally end up in a "Zombie" channel
and stop receiving calls. (Note that the console never says "Zombie" it
just shows a channel that can't be hung up.
Here's an excerpt from a console
Trying to compile app_swift with Asterisk 14.2.1 and getting the
following. Can anybody tell me what I'm missing?:
[root@localhost app_swift-master]# make
____
(_) / __) _
_
?
Thanks,
*Brent Davidson*
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, the reverse
lookup query failure caused the delay around(7-9 seconds). The purpose
of reverse lookup is to block IP Spoofing attacks.
Regards,
Faheem
On Tue, Jun 7, 2016 at 7:48 PM, Brent Davidson
<br...@texascountrytitle.com <mailto:br...@texascountrytitle.com>> wrote:
In trying to troubleshoot the Delay after Answer problem I had before
(which seems to be fixed), I have somehow created a new problem:
Outgoing calls are now failing with the following message:
[Jun 7 13:28:09] WARNING[9247][C-]: app_dial.c:2429
dial_exec_full: Unable to create
,
Brent Davidson*
*
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asterisk-users mailing
I'm trying to get app_swift (app_swift-2.1-b1-ast10) working with
Asterisk 10.2.1 on Ubuntu Server 11.10. Everything appears to compile
correctly, but when I go to load the module I get the following:
server*CLI module load app_swift.so
Unable to load module app_swift.so
Command 'module load
On 4/12/2012 3:09 PM, Patrick Lists wrote:
On 04/12/2012 09:09 PM, Brent Davidson wrote:
I'm trying to get app_swift (app_swift-2.1-b1-ast10) working with
Asterisk 10.2.1 on Ubuntu Server 11.10. Everything appears to compile
correctly, but when I go to load the module I get the following
Well, I was wrong. The messages went away for a day, then came back. I
am now rebuilding the server using an older motherboard. Hopefully that
will solve the problem.
On 12/9/2011 4:09 PM, Brent Davidson wrote:
For the sake of posterity, I'm posting this solution:
When I checked
For the sake of posterity, I'm posting this solution:
When I checked the server, the PnP OS option in the BIOS was set to
No. Changing the option to Yes and rebooting has solved the problem.
On 12/8/2011 10:58 AM, Brent Davidson wrote:
I am still having issues with the error message
Dec
I am still having issues with the error message
Dec 7 14:25:06 servername kernel: FXO PCI Master abort
filling up my log files. I've temporarily managed a work around by
having the message log emptied every 10 minutes, but this is not a
permanent solution.
I expanded my google search to
I am running Asterisk 1.6.2.20 with dahdi 2.5.0.2, Oslec from kernel
sources. Hardware is 2 X100P Wildcards. Everything seems to be working
OK but my logs are filling up with this message:
Dec 7 14:25:06 servername kernel: FXO PCI Master abort
The messages just pour in constantly until the
...@lists.digium.com] On Behalf Of Brent Davidson
Sent: Wednesday, December 07, 2011 2:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Help! Logs filling up with errors!
I am running Asterisk 1.6.2.20 with dahdi 2.5.0.2, Oslec from kernel
sources. Hardware is 2
--
Brent Davidson
Texas Country Title Company
112 W 2nd / P.O. Box 663
Cameron, TX 76520
254-605-0140 ex. 21
br...@texascountrytitle.com
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On 5/26/2010 1:16 PM, Tim Nelson wrote:
- Jeff LaCoursierej...@jeff.net wrote:
On Wed, 26 May 2010, Brent Davidson wrote:
Just set the POTS lines to answer after a second ring rather than
after
the first. Problem solved.
Now that sounds like a good
On 4/7/2010 2:45 AM, asterisk card support wrote:
hi:
how about the codecs?
Best wishes!
Asterisk Support group(sangoma, digium...), providing asterisk conf,
pri, ss7, elastix, trixbox support.
website:www.cnasterisk.com, www.voip88.com
I have the phones and asterisk limited to ulaw
On 3/31/2010 10:38 AM, Michael L. Young wrote:
Is there a chance that you are using Realtime at all?
I am just curious because I was having problems with dropped calls as well
and just discovered that it appears to be related to the database server.
If for some reason on the database server
On 3/31/2010 12:06 PM, Danny Nicholas wrote:
Just to get a 100% correct response to last question, are you using the flat
CDR or mysql/some other DB?
All sip clients/peers are defined in sip.conf, dial-plan is entirely in
extensions.ael. We have one office that uses an Asterisk native
On 3/31/2010 12:16 PM, Philipp von Klitzing wrote:
Maybe rtptimeout in sip.conf is involved (and not behaving as expected)?
I was suspecting something with either rtptimeout or sip registration
timeout, but I'm not sure what.
--
, but on the
off chance that my logs catch either a drop or a one-way audio, the sip
debug looks like just a normal call.
Is there any setting that might cause both one-way audio and dropped calls?
Thanks,
Brent Davidson
On 3/30/2010 3:14 PM, Danny Nicholas wrote:
A few thoughts;
1. I assume that the * servers aren't on dedicated networks; Do the dropped
or one-way calls occur during high-traffic times or are they concurrent with
large downloads? In my shop, we had to get a router that would prioritize
that both problems may be related though. Possibly a
registration issue? Any ideas are welcome.
Thanks,
Brent Davidson
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Is there a way to detect if a call is a transfer in the dialplan? Here
is my issue: I have an office with 2 extensions. Under normal
circumstances any call that comes in should ring both extensions. I
accomplish this through a queue. The problem is that if the call is
answered on say
Anybody seen this article yet? Looks like Russian Telecom business have
decided that VoIP is going to put a dent in their profits so their
pitching it as a threat to Russia's national security and working to get
laws put into place to make sure the government controls VoIP providers
operating
Gondar Monn wrote:
I am having trouble with a DID on a PRI. If there is a call to
that DID (let say 5551234) , the next calls get a busy signal. How to
I go about sending the call to the next available channel ?
Thanks!
G.
If the telco is providing the PRI then you need to tell
Alex Samad wrote:
Hi
I have setup forwarding - xfering - where you press # and then the
extension. I add t to the dial cmd.
My problem is that when you call something like internet banking they
want #, but when # is pressed asterisk gets it instead. is there a way
around this ?
I haven't
Danny Nicholas wrote:
If you are using a large number of DAHDI channels, you could designate a
chunk of them as non-local since you can control RXGAIN on each channel.
You would have to work out something with your TELCO since your'e a dead
duck control-wise once you answer the call.
Is there any possibility of DAHDI supporting Automatic gain control on
TDM ports? I'm having issues at a couple of offices where calls made to
local numbers are fine but a when a calls from or goes to a large
percentage of long-distance or 1-800 numbers the person at the remote
end cannot
Julien Claassen wrote:
Hello!
I've configured Music on Hold in asterisk, the only, most certainly,
stupid
problem I have is, which DTMFs to send to activate and deactivate it.
If I use the cli, I can establish a call with originate. With the misdn
send digit command I can send a
Botond Botyanszki wrote:
Hi,
I have an x100p zaptel card with asterisk 1.4. I'm using the system for
outgoing calls.
My problem is that Answer() is falsely returning while the call is still
ringing and was not really answered yet. I've been digging google, wikis
but have not found what
Steve Totaro wrote:
On Thu, Jun 18, 2009 at 12:46 PM, Brent Davidson
br...@texascountrytitle.com mailto:br...@texascountrytitle.com wrote:
John A. Sullivan III wrote:
Hello, all. I am delightfully slogging my way through installing and
configuring Asterisk 1.6.1.1 on CentOS
John A. Sullivan III wrote:
Hello, all. I am delightfully slogging my way through installing and
configuring Asterisk 1.6.1.1 on CentOS 5.3. I'm learning lots and
admiring the product but I'm having a problem getting speex to install
and I would very much like to use it. It is not available
Have you tried relaxdtmf=yes in zapata.conf/dahdi.conf?
-Brent
Timm M.Schneider wrote:
Hi,
is there a possibility to tell zaptel or Asterisk to modify the DTMF
sensibility?
The problem what i have is that the Asterisk don't get all Numbers which the
analog-FAX dial, let say the FAX dial
Administrator TOOTAI wrote:
David fire a écrit :
out there is a file to change the dtmf duration
where are you?
France
[...]
from other phones like lkand lines it works well?
No, the same. The called number is a number received by a trunk SIP, the
GW is also setted as
David Backeberg wrote:
On Thu, May 7, 2009 at 3:54 PM, Brent Davidson
br...@texascountrytitle.com wrote:
I've got multiple satellite office all linked back to the main office
via VPN. Each office has their own asterisk server which registers back
to the main office's Asterisk server. Each
Jeremy Mann wrote:
Access-list 100 permit ip host asterisk server any
Class-map match-any voip
Match access-group 100
Policy-map voip
Class voip
Priority 256
Class class-default
Fair-queue
Interface fastethernet 0
Service-policy output voip
Above is what I do to prioritize
boxes to unnecessary security risks? (At present all of our asterisk
boxes are behind the firewalls and only talk to each other over the
VPN. All PSTN connection is done through TDM boards so they have no
direct exposure to the internet.)
Thanks,
Brent Davidson
Alternatively look into the M() option to Dial to execute a Macro upon
connect. You could have your macro setup to call the cepstral app.
-Brent
Justin Killen wrote:
That works great -- Thanks Danny!
-Justin
that be
better as far as the duration is concerned?
on Monday 04/13/2009 Brent Davidson(br...@texascountrytitle.com) wrote
John covici wrote:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to increase the duration of the tone, its pretty
short
It's been around awhile. I've used it in 1.4 Check out this link for
basic info: http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPdtmfmode
John covici wrote:
Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4.
Is this new in 1.6?
on Tuesday 04/14/2009 Brent Davidson(br
Danny Nicholas wrote:
Do you have include=intern in the default context? If no, * will come
back with can't find peer 210 (or 211).
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *jonas
kellens
*Sent:* Monday, April
John covici wrote:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to increase the duration of the tone, its pretty
short and some IVR's are unhappy or don't detect it. I did poke
around, but it looks like when RFC2833 is used, it actually generates
rtp
Jorge Mendoza wrote:
Are there an IOS interface for Asterisk?, or an IOS to SIP converter?
Some femtocells uses this protocol and I would to use them with Asterisk.
Jorge Mendoza
___
You're comparing to apples to Orange. IOS is the Cisco
Jorge Mendoza wrote:
Brent Davidson wrote:
Jorge Mendoza wrote:
Are there an IOS interface for Asterisk?, or an IOS to SIP converter?
Some femtocells uses this protocol and I would to use them with Asterisk.
Jorge Mendoza
Cary Fitch wrote:
It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8
bits each, and that is 2^140^8, a nearly inexhaustible key number which is
related to audio and video data simultaneously stored on a Google Database,
which is then sent to the user.
Thus with the 140
Both Montior and MixMonitor are part of the standard Asterisk
distribution. There is no need to download anything else.
bilal ghayyad wrote:
Thsi Monitor CMD is a part of AsteriskNow so I have to download AsteriskNow and
then take only the Monitor CMD or I can download the Monitor CMD alone?
David Backeberg wrote:
On Sat, Mar 14, 2009 at 12:00 AM, Steve Underwood ste...@coppice.org wrote:
Fully open-to-the-public FAX servers tend to get just get a lot of bad
calls, many of them wrong numbers, or voice users. FAX servers for
I've definitely seen that, and have been able to
1246463 is not the same as 246463. Note the missing 1
If you want to match what is being dialed then your extensions.conf
should look like this:
[default]
exten = 246463,1,Answer(SIP/8003)
Bayardo Sanchez wrote:
in my extension.conf i set :
[default]
exten =
Danny Nicholas wrote:
Greetings listers,
I am running Asterisk 1.4.21.2 on Suse 11.0
on a Dual Processor Dell Poweredge 1650. I recently attempted to
update the BIOS and now have this happen:
When the machine starts up, Asterisk runs fine. When I do a large
amit mehta wrote:
Hello Members,
Sorry for hijacking the earlier thread and asking the question last time.
Is anyone aware about a solution to call incoming number and dictate
the files by using Dictate feature of Asterisk used for Medical
Transcription industry.
Thanks Regards,
Robert Broyles wrote:
Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
--
Regards,
Robert Broyles
Eric Wieling, Asteria Solutions Group wrote:
Robert Broyles wrote:
So I'm using the READ() application within an IVR, and having a strange
issue, and
to be
there, but they show up anyway.
Can someone else check this on their system, and see if this is a problem?
--
Regards,
Robert Broyles
Brent Davidson wrote:
Robert Broyles wrote:
Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
--
Regards,
Robert Broyles
Eric
You need canreinvite=no in the config for your sip phone and the
veracity connection, otherwise Asterisk will just mediate the call setup
then try to allow the sip phone and veracity to talk directly to one
another.
Jim Dickenson wrote:
I have a SIP phone at home behind a NAT router
Jerry Geis wrote:
hi,
try to set the rtptimeout value in sip.conf to a resonable value - so
asterisk will kill the channels if it does not receive rtp traffic for
the specified time
regards,
Wolfgang
I uncommeted the rtptimeout=60 value in sip.conf and did a reload.
It still hasnt
Look int the ChannelRedirect command.
Geoff Lane wrote:
Hi All,
I'd appreciate some help on how to implement call stealing. That is,
where you dial a code to redirect any call on the system to your
handset.
I'm getting rid of my BRI service and I'm trying to replace the
functionality of
Watkins, Bradley wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent
Davidson
Sent: Wednesday, December 31, 2008 1:03 PM
To: m...@digium.com; Asterisk Users
Benoit wrote:
Brent Davidson a écrit :
Another question along these lines... If I set a Global called
TRUNK in the globals section and later assign do a TRUNK=whatever it
appears that a local variable called TRUNK is created instead of using
the global. You must explicitly use the Set
Steve Murphy wrote:
On Tue, 2008-12-23 at 12:14 -0600, Brent Davidson wrote:
I have two offices sharing a phone system. They also share a common
internal context because all of the employees of the second office also
work for the first office. Each office has 4 outside lines and I have
On my asterisk system, if an incoming call only has a number for the
caller ID and no name, the system is using the channel name as in the
Callerid Name field. I would like to use some sort of pattern match
test to test for the presence of Zap/ in the ${CALLERID(name)}
variable and if it is
I have two offices sharing a phone system. They also share a common
internal context because all of the employees of the second office also
work for the first office. Each office has 4 outside lines and I have
defined 2 channel groups in my zapata.conf. The second office needs all
of their
Philipp Kempgen wrote:
Brent Davidson schrieb:
On my asterisk system, if an incoming call only has a number for the
caller ID and no name, the system is using the channel name as in the
Callerid Name field. I would like to use some sort of pattern match
test to test for the presence
Philipp Kempgen wrote:
Brent Davidson schrieb:
macro outside-dial ( num ) {
if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK=Zap/r2;
} else {
TRUNK=Zap/r1;
}
Dial(${TRUNK}/${num},,Ttok);
}
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning
Dave Fullerton wrote:
I had gotten similar messages when I forgot to put quotes around
channels like that (took me forever to realize that one). Since you have
them I would say this is a bug. What version of asterisk are you running?
-Dave
I'm running 1.4.21.2 and I can't upgrade until
Tzafrir Cohen wrote:
On Tue, Dec 23, 2008 at 03:09:51PM -0600, Brent Davidson wrote:
Unfortunately 1.4.22 no
longer has Zaptel. :(
Asterisk 1.4.22 builds with both Zaptel and DAHDI.
I spent several hours trying to make it work yesterday and it just
wouldn't. I kept getting
Jeff LaCoursiere wrote:
On Tue, 23 Dec 2008, Brent Davidson wrote:
Dave Fullerton wrote:
I had gotten similar messages when I forgot to put quotes around
channels like that (took me forever to realize that one). Since you have
them I would say this is a bug. What version of asterisk
Tzafrir Cohen wrote:
What error message from where?
With Zaptel the echo canceller settings are global (that is: one
hard-coded echo canceller). With DAHDI there are echo canceller modules
and you can (and actually need to) set them per-channel.
It might have something to do with the
Tilghman Lesher wrote:
On Monday 15 December 2008 00:57:08 Langdon Stevenson wrote:
Hi Paul
Thanks for the reply. I have removed and re-installed all of the Fedora
Zaptel packages with Yum. I have the following installed:
asterisk-zaptel 1.4.12.1-1.fc8
zaptel.i386
I have several branch offices all running Asterisk PBX's that register
to each other via SIP so that calls can be transferred from office to
office. Everything is working great on the office to office transfers,
but I'd like to somehow make the CallerID more useful. Currently if an
extension
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
-Dave
There aren't any callerid= entries in any of my sip peer entries, and
Dave Fullerton wrote:
Brent Davidson wrote:
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
-Dave
There aren't
John Todd wrote:
Erik -
Have you found RealSpeak to be worth the cost? Can Cepstral, with
the hourly $ spent on tuning, be made to be a reasonable substitute?
It's been a while since I did a head-to-head comparison between
Cepstral and (anything else) so I did a quick demo of the
Mikel Lindsaar wrote:
This must be how the Telco actually managed to router the call.
Because it must go 'pri signaled digits first, inband second'.
Because if you take the pri signal digits (which we assume are the
first three) and put them at the start, you can see the number, all in
Try flushing all of your iptables and see if that helps. See if there's
anything in your dmesg that might indicate what's up.
Jeff LaCoursiere wrote:
Sorry again for the only marginal relation to asterisk, but the issue does
affect the voice performance I am experiencing, so I am soothing my
Have you verified that the NTP server has the correct time? Also, if
you're grabbing the time from a source set to GMT you'd need to set the
gmtOffset field.
Doug Smith wrote:
Tried to submit this email this morning and didn't see it in the
list. I apologize if it is a dupe.
I've
I have a weird thought... Is the PBX possibly passing the digits both
inband and via PRI signaling so Asterisk is getting two digit streams at
the same time and totally freaking out?
Mikel Lindsaar wrote:
I plug the NEC back straight to the Telco and all works well again.
I just got
Jerry Geis wrote:
I have a small system, server, client and 2 phones. Phones are polycom
501's.
In general all is working fine. I can call the two phones, speak etc...
I can have the server call each phone and play a wave file.
However, when trying to setup a direct dial number of 1044 that
Jerry Geis wrote:
Are your polycom phones set up for overlap dialing or do you dial the
number then press a key to dial?
From you message I tried a couple things...
Clicking New call, then starting to dial this is when it messes up.
when I start entering the number first then click
If I have a global variable in my dialplan and I change it, does that
change immediately take affect for all calls that are active?
Here is my situation. The company I work for has two office groups that
share a building. The two offices are separate companies but support
one another and
Tilghman Lesher wrote:
[companyA]
exten = _X.,1,Set(company=A)
exten = _X.,n,Goto(maincontext,${EXTEN},1)
[companyB]
exten = _X.,1,Set(company=B)
exten = _X.,n,Goto(maincontext,${EXTEN},1)
I should probably also mention that I am using AEL for my dialplan.
(i'm a programmer and the
Singer X.J. Wang wrote:
He's dead, if you look at the recent photos of him his shadow is not
where it should be compared to other people in the photos.
Well that's just lovely. Kim Jong Il is now an immortal vampire.
Better call the white house and tell them to replace the nuclear warhead
There is going to be a bit of a current output limit on the FXS card.
For the actual limit you will need to contact the manufacturer. Phones
that use digital ringers will be much more likely to work than phones
that use mechanical ringers.
Mike wrote:
Folks,
I have a TDM400 with an FXS
I ran into almost this exact same problem when I first installed
asterisk. My company uses a virtualdomain hosted by our isp. We'll
call it mycompany.com for example. When I first set everything up I
wasn't able to send any mail from the asterisk server even though it was
on an accepted IP.
for these phones, they will trust the sip header for IP
address and may misroute.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson
Sent: October 24, 2008 7:36 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Sporadic One Way Audio
Importance: High
, but there is no firewall between the asterisk server and the
phones and no iptables or anything like that running on the Asterisk
server and sifting through sip debug logs to try to find one call out of
maybe 50 has so far proven fruitless.
Are there any common issues that might cause this?
Thanks,
Brent Davidson
connected to the PSTN? SIP/IAX out...ISDN/T1
out? Etc...
Are you looking for lost RTP between * and internal phones or * and external
provider?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson
Sent: October 24, 2008 5:55 PM
To: Asterisk
GNUbie wrote:
What particular configs are you looking for? Below is my current setup
and scenario:
[snom] ==LAN== [asterisk] ==FXO/POTS == [analog_telephone/mobile_phone]
SNOM is using the 192.168.101.102 IP address
Asterisk is using 192.168.101.1 IP address for its eth1 interface
FXO port
Brent Davidson wrote:
Babcock, Michael Alex wrote:
hey;
i'm at best western and am curious is there a way i could find out if
our best western, with out asking, is using asterisk?
oh and petsmart i think is using asterisk they have alason voice for
there main voicem enu.
mike
thanks
I'm trying to test out Speex for our branch to branch connections, but
am running in to a problem. I downloaded the Speex source code for
1.2rc1, did a ./configure, make and make install then went to my
asterisk folder did a ./configure, make clean make menuconfig verified
that speex is
Brent Davidson wrote:
I'm trying to test out Speex for our branch to branch connections, but
am running in to a problem. I downloaded the Speex source code for
1.2rc1, did a ./configure, make and make install then went to my
asterisk folder did a ./configure, make clean make menuconfig
Daniel Hazelbaker wrote:
On Oct 9, 2008, at 2:59 PM, Brent Davidson wrote:
Short answer: currently no.
Medium answer: I just rolled out 60+ Snom phones (300s and 320s) and
we do call parking with DTMF. People were used to just hitting PARK
and their phone displaying the park
Doug Lytle wrote:
Brent Davidson wrote:
Also be aware that in 1.2.x and 1.4.x, if you park a call and then
pick it up, you can't park it again. At least not with the DTMF
I wasn't aware of the inability to re-park calls in 1.4 That could
have been a nasty surprise. I would
Doug Lytle wrote:
Brent Davidson wrote:
Ok, the patch is working great. Any idea what would make the one step
parking not work? I've tried several DTMF combinations in features.conf
Check your featuredigittimeout, it defaults to 1/2 second. You may need
to increase it.
I
Daniel Hazelbaker wrote:
You won't. The patch I sent you off-list is incomplete, this one is
better. I forgot I fixed the parked has timed out option in another
patch before I fixed this part. Anyway, make sure when you dial you
put k in the dial options (K too if you want both sides
Babcock, Michael Alex wrote:
hey;
i'm at best western and am curious is there a way i could find out if
our best western, with out asking, is using asterisk?
oh and petsmart i think is using asterisk they have alason voice for
there main voicem enu.
mike
thanks for reading
Systems
of the tones, which is annoying.
Is there any way to set up the transfer silently and still get the
parking slot extension back?
Thanks,
Brent Davidson
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