about 1 call in 5. I'm wondering if
it's just because they call us more than any of our other customers or
if there is some peculiarity with their phone system. Anybody have any
ideas what to try next?
Thanks,
Brent Davidson
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an extension about 1 call in 5.
I'm wondering if
it's just because they call us more than any of our
other customers or
if there is some peculiarity with their phone system.
Anybody have any
ideas what to try next?
Thanks,
Brent Davidson
because they call us more than any of our
other customers or
if there is some peculiarity with their phone system.
Anybody have any
ideas what to try next?
Thanks,
Brent Davidson
Looking for last
I seem to be having similar problems at one of my branch offices. See
my message on Intermittent DTMF issues for some of the standard
replies. Have you tried the RelaxDTMF tag in zapata.conf? I don't
think Gain calibration applies to T1 cards or I would recommend that as
well.
Thanks,
Do you have canreinvite=no in the sip client configuration? If not then
the two sip phones are probably issuing a reinvite command and taking
asterisk out of the call path. If that happens and the phones can't
reach consensus on a codec then you run into audio problems. If you're
not a
Does anyone know of a way to make a Snom 300 phone monitor the parking
lot extensions and allow one-button pickup with the programmable buttons?
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To
I've been reading most of the day and can't seem to find a clear
definition of the syntax for parking lot hints in AEL2. I have tried
all of the following and they either do not light up the line button on
my Snom 300 or give syntax errors:
hint(park/701) 701 = {
ParkedCall(701);
}
to further relax the
DTMF detection?
Thanks,
Brent Davidson
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system, and it is not limited to any one caller.
Thanks,
Brent Davidson
Brent Davidson wrote:
Still grasping at straws trying to solve DTMF detection issues with one
of my asterisk servers. This particular server is now running Asterisk
1.4.18.1 and Zaptel 1.4.9.2 in runlevel 3 (console
LOL
Reminds me of that old Ray Stevens Song - Jeremiah Peabody's
Polyunsaturated Quick Dissolving Fast Acting Pleasant Tasting Green and
Purple Pills
Oh Yeah Binary System = Pyramid Scheme
BJ Weschke wrote:
I'll give you an A+ for originality after I get done laughing and then
audio examples to anyone that is interested.
Thanks,
Brent Davidson
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With canreinvite=no you are forcing asterisk to remain in the call
path. As long as Asterisk is in the call path, it is supposed to be
transcoding the calls, so it doesn't care what the compatible codecs are
between then endpoints. Each leg of the call is phone-asterisk so
asterisk
You could also, conceivably, handle this outside of asterisk by using a
more complex MOH stream source. For instance, use a shoutcast client as
the MOH source, run your own shoutcast server streaming your music and
have a script set up to periodically interrupt the stream being served
to the
Greg Woods wrote:
Shaun Ruffell wrote:
svn co http://svn.digium.com/svn/zaptel/branches/[EMAIL PROTECTED]
zaptel-1.4-4122
Thank you, I will try that tonight when I get home and report back.
--Greg
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the MOH to ringing and the
majority of the complains stopped. (The remaining complaints are
related to DTMF detection problems.
Just food for thought.
Good luck,
Brent
Atis Lezdins wrote:
On Wed, Apr 2, 2008 at 11:05 PM, Brent Davidson
[EMAIL PROTECTED] wrote:
You could also, conceivably
Asterisk Development Team wrote:
The Asterisk.org development team has announced the release of Zaptel
versions 1.2.25 and 1.4.10. These releases contain many bug fixes as
well as performance enhancements.
A couple of the more major changes include: modifications to the
wctdm24xxp and
Jason Parker wrote:
Brent Davidson wrote:
Do they mean 1.4.20 instead of 1.4.10? If not, then this message was
seriously delayed :-D
-Brent
Zaptel, not Asterisk. :)
1.4.10 is correct.
Doh! My bad.Was looking at the wrong version numbers. As many
times as I've
Have you tried the using the SIPDtmfMode function in your dial plan?
It can be used to change the DTMF mode between two points in a call.
The problem, I would think, would be if your phones are set up to ONLY
send inband audio then you have to find someway to get audio to
transcode the DTMF
this problem or have any ideas how to
eliminate it?
Thanks,
Brent Davidson
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Brian J. Murrell wrote:
On Wed, 2008-04-09 at 17:21 -0500, Brent Davidson wrote:
Does anyone know if Asterisk will convert an inband DTMF from one sip
channel to an info or rfc2833 DTMF on another (i.e. bridged) SIP
channel?
You might also try canreinvite=no for both your phone
Jon Pounder wrote:
I had the phantom rings for years, once a day same time roughly every
day, finally just got annoyed enough one day I trapped the telco on
the phone with me till I finally got to talk to the right person. The
right person knew instantly what I was talking about after
Brian J. Murrell wrote:
On Thu, 2008-04-10 at 11:13 -0500, Brent Davidson wrote:
You might also try canreinvite=no for both your phone and the sip
peer.
Yeah, there is definitely no re-inviting going on. Both Asterisk and
the local handset are in a local network behind NAT
I have an operator queue that is supposed to ring 2 phones, extension 10
and 11. Everything is working correctly but I keep seeing these
messages in my log: The device state of this queue member, Sip/10, is
still 'Not in Use'. Everything I've been able to find on this message
so far points
Lee Jenkins wrote:
Brent,
I had a similar problem and I feel for you, its frustrating.
Are you using polycom phones by chance? Here is the problem that I had, not
sure if your problem is related.
Specs:
- 6 Polycom 301 phones.
- CentOS 4 Server with Asterisk 1.2.x
- Sangoma A200 card
Alex Balashov wrote:
Greetings,
This may have already been asked many times, but I cannot seem to find a
satisfactory and consistent answer anywhere.
I have an X100P card (from x100p.com) installed in a Dell PowerEdge 2850
or 2650 (cannot recall):
00:00.0 Host bridge: Broadcom CMIC-WS
Tzafrir Cohen wrote:
On Wed, Apr 16, 2008 at 09:44:25AM -0500, Brent Davidson wrote:
Alex Balashov wrote:
Greetings,
This may have already been asked many times, but I cannot seem to find a
satisfactory and consistent answer anywhere.
I have an X100P card (from x100p.com) installed
John Signorello wrote:
excuse me...
But did you not just post
[asterisk-users] OT Nice IBM 1U Server Gets Along w/Old and New
Digium Boards Cheap X305 $199
Did you not provide a link to a COMMERICAL entity?
Wasn't your a post a unsolicited post, that is, not in response to a
Jerry Geis wrote:
The netstat show 0.0.0.0
netstat -anp | grep :69
udp0 0 0.0.0.0:69
0.0.0.0:* 4007/xinetd
--
cat /etc/xinetd.d/tftp
# default: off
# description: The tftp server serves files using the
Jerry Geis wrote:
Brent, below is the file. Looks good to me... Also Both networks start
at boot. Nothing is manual on this box at all.
--
# Simple configuration file for xinetd
#
# Some defaults, and include /etc/xinetd.d/
defaults
{
instances
A friend of mine recently told me about a phone system his office was
considering that did not use any handsets. Instead of a phone, each
person was issued a BlueTooth headset and several bluetooth repeaters
were installed throughout the building. When a call came in, it would
be routed to
interrupting the call.
For smaller setups there is 3Com WXR100 that supports up to 3 MAPs (Managed
Access Points).
AttVinícius FontesDesenvolvimentoCanall Tecnologia em Comunicações Ltda.
- Brent Davidson [EMAIL PROTECTED] escreveu:
A friend of mine recently told me about a phone system his
Which phones are you using and what software revision. I've had a crash
course in Snom phone lately and can probably help with at least the park
orbits.
-Brent
Thermal Wetland wrote:
I would like to hire someone to help us tweak our asterisk system for
Snom phones.
We would like to
button as type DTMF with that key sequence
assigned to it (in this case *1). (There may be another way, but this
seems simplest.)
Let me know if you need any more help.
-Brent
Thermal Wetland wrote:
On Thu, May 8, 2008 at 5:20 AM, Brent Davidson
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED
Another solution that works for me is to add Playback(silence/1) just
before whatever you are about to do. Something about the playback
command opens the channel up.
-Brent
Sherwood McGowan wrote:
Alan Lord wrote:
Sherwood McGowan wrote:
snip /
Hrm...I have
that
is not an option.
Given this setup, is there any reason for me to switch to Asterisk 1.6
or should I stick with 1.4?
Thanks,
Brent Davidson
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Tzafrir Cohen wrote:
On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote:
Brent Davidson wrote:
We're currently using Asterisk 1.4.19, Zaptel 1.4.10,
Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones.
Why on earth are you running two layers of echo cancellation
Matt Watson wrote:
Have you tuned rxgain txgain in Zapata.conf? shameless-self-plug
http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/
/plug
Also, have you used fxotune to tune each FXO interface?
I believe echo cancellation happens at the Zaptel /
Just an update. I tried updating to the newest Rhino Release firmware
1.15 and newest stable driver version 2.2.6. It works OK with
zaptel-1.4.9.2 and compiles OK with 1.4.10.1 but when compiled against
zaptel 1.4.10.1 Asterisk does not see any zap channels. I'm currently
running one branch
Philipp von Klitzing wrote:
Hi!
I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We
are using asterisk 1.4.2 for a SIP only based configuration. [...] We
are planning to accomodate about 5,000 users on this server.
Many people on this list will advise you to use a
Correct me if I'm wrong, but unless you pass specific options to the
dial command to have it override the ringing then when you dial out, you
hear the audio from whatever channel you're dialing on. So the tones
you are hearing are from the telco. The ring cadences defined in
indications.conf
I'm using the Rhino R4FXO cards to answer incoming voice calls and just
had this idea last night... Is there any type of module for asterisk
that will divert a call to a softmodem? For example, I call in on the
voice lines, get the main menu, instead of dialing an extension to get a
person I
Steve Totaro wrote:
On Wed, Jun 11, 2008 at 1:47 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Wed, Jun 11, 2008 at 11:53 AM, Raj Jain [EMAIL PROTECTED] wrote:
On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell [EMAIL PROTECTED] wrote:
I'm wondering if the SIP lines can start
Steve Totaro wrote:
On Wed, Jun 11, 2008 at 2:44 PM, Brian J. Murrell [EMAIL PROTECTED] wrote:
If you ever have problems with a call dropping after 30 seconds,
Answer() is usually the cause.
Thanks,
Steve T
This is the first I've heard of this. I've never actually had the drop
after
And there are people like me who still can't get PRI's for less than
$1100/month. (Granted, I doubt I'll ever need a pri for the business I
am with now, but I was with an ISP for a long time that still supported
dial-up and we had 8 PRI's with a bulk discount that got them for us at
bkruse wrote:
Yes, probably, same basic error.
-brandon
Fidel Garcia wrote:
Great info! Thanks!
However, they do not mention the fact that when you create a new user you
cannot select the DialPlan. I wonder if the path fixes both issues. Any
idea?
Fidel Garcia
System Engineer
Robert Goodyear wrote:
Yeah I'm thinking either homeland security or some other
identity-critical legislation might be on my side here.
On Thu, Jul 3, 2008 at 12:40 PM, randulo [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
On Thu, Jul 3, 2008 at 9:26 PM, Robert Goodyear
[EMAIL
I have a question about click to dial. Each of my users is going to
have a VOIP phone with an assigned extension. Is there a simple way to
build a web-based speed-dial list that will allow them to put in their
extension, click on the number they want to dial, and have asterisk ring
their
So you basically want a call-interrupt feature that puts the interrupted
party on hold?
rachid wrote:
Hi,
I want to make an insertion in a communication; A et B are in
communication, an other C wants talk to A, how can i set B on
hold state and make a call to A?.
Thanks.
Rachid
Doug Lytle wrote:
Eric ManxPower Wieling wrote:
But what would you call it? It's not a card, so it can't be a NIC, right?
(N)etwork (I)nterface (C)ontraption
Doug
(N)etwork (I)nterface (C)onnector
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Steve Totaro wrote:
On Tue, Aug 19, 2008 at 7:10 AM, Jim Boykin [EMAIL PROTECTED] wrote:
We run asterisk to handle incoming DIDs and we have observed
inefficient Codec Translation.
Here is the scenario
[DID Vendor] --- [Asterisk ]
External
Manolet Gmail wrote:
Have, i want to create a sip extension to a context in my dialplan.
how i can do that?
___
Simple. Use a Goto:
[context1]
exten = 123,1,Goto (context2,456,1)
[context2]
exten = 456,1,Background(tt-monkeys)
Steve Murphy wrote:
On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote:
Greetings,
i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox..
i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls
Julian Lyndon-Smith wrote:
I am trying to track a strange bug down, and need to ask a really stupid
question, just so I can eliminate the possibility ..
When a SIP channel is hung up, I import a variable called MEETMEROOM
from the BRIDGEPEER channel, and if it is set, jump to another part
?
Thanks,
Brent Davidson
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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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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
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
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
, 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
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
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.
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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